From velu.technical at gmail.com Tue Jun 1 02:50:29 2010 From: velu.technical at gmail.com (velusamy Krishnan) Date: Tue, 1 Jun 2010 15:20:29 +0530 Subject: [Freeswitch-users] Read and Set the UUI in PRI Message-ID: Dear All, How do I set and read the UUI(User-User-Information) in FreeSWITCH? Thanks, Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/35642546/attachment-0001.html From tculjaga at gmail.com Tue Jun 1 04:41:36 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 1 Jun 2010 13:41:36 +0200 Subject: [Freeswitch-users] time format Message-ID: hello, I need to convert seconds into hh:mm:ss format, any hint ? T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/959c20f5/attachment.html From jaybinks at gmail.com Tue Jun 1 04:49:05 2010 From: jaybinks at gmail.com (jay binks) Date: Tue, 1 Jun 2010 21:49:05 +1000 Subject: [Freeswitch-users] time format In-Reply-To: References: Message-ID: my hint : how / what are you trying to do this in ?? LUA, PHP, PERL, C# ? a little bit more info would help. J On Tue, Jun 1, 2010 at 9:41 PM, Tihomir Culjaga wrote: > hello, > > I need to convert seconds into hh:mm:ss format, any hint ? > > > T. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100601/e2177bfa/attachment.html From tculjaga at gmail.com Tue Jun 1 06:39:22 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 1 Jun 2010 15:39:22 +0200 Subject: [Freeswitch-users] time format In-Reply-To: References: Message-ID: On Tue, Jun 1, 2010 at 1:49 PM, jay binks wrote: > my hint : > how / what are you trying to do this in ?? > > LUA, PHP, PERL, C# ? > > a little bit more info would help. > > J > ahhh sorry, forgot to mention .. > im trying to do it in dialplan .... i have a channel variable holding the time e.g.: phrase for timeleft expects hh:mm:ss time format .... any clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/d59009c0/attachment.html From lists at infosecurity.ch Tue Jun 1 06:41:17 2010 From: lists at infosecurity.ch (Fabio Pietrosanti (naif)) Date: Tue, 01 Jun 2010 15:41:17 +0200 Subject: [Freeswitch-users] What happen when leg-a & leg-b are on SIP/TCP and one get a socket disconnection during a call? Message-ID: <4C050DFD.8080708@infosecurity.ch> Hi, sorry for using the so big subject, i am experiencing some strange behaviour i am going to enter more in depth debugging in the next weeks. I have the following situation: - A call getting established - both peers are using SIP/TLS over TCP - leg-b get disconnected at TCP level (such as WiFi disconnection or just a TCP hard reset by a NAT device reloading it's rules) In such condition i noticed, but still need detailed investigation, that FS does not detect that leg-b TCP socket died, thus notifying immediately leg-a that the call cannot be completed. FS instead, it seems from log/tcpdump observation, wait until the various SIP timer expire in order to provide back to leg-a error that the call cannot be completed. If it's that way, it would be a nice improvement, when a SIP/TCP or SIP/TLS client get a TCP disconnect, to immediately react on the other leg of the call by notifying the proper SIP error, because the layer4 (SIP) connection is lost due to a layer3 (TCP) connection break. Is this something feasible/reasonable? Fabio From david.ponzone at gmail.com Tue Jun 1 07:50:29 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Tue, 1 Jun 2010 16:50:29 +0200 Subject: [Freeswitch-users] time format In-Reply-To: References: Message-ID: <78B6C072-C1D4-4270-8AB0-C355984DD13D@gmail.com> Tihomir, you can do that yourself with expr. h = expr floor(credit_time/3600) m = expr floor(mod(credit_time,3600)/60) s = expr mod(credit_time,60) Just convert that to the right $expr{} syntax. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 01/06/2010 ? 15:39, Tihomir Culjaga a ?crit : > > > On Tue, Jun 1, 2010 at 1:49 PM, jay binks wrote: > my hint : > how / what are you trying to do this in ?? > > LUA, PHP, PERL, C# ? > > a little bit more info would help. > > J > ahhh sorry, forgot to mention .. > > im trying to do it in dialplan .... > > > i have a channel variable holding the time > > > e.g.: > > > > > > > > > phrase for timeleft expects hh:mm:ss time format .... > > > any clue? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/f907db06/attachment.html From andrew at hijacked.us Tue Jun 1 11:43:15 2010 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 1 Jun 2010 14:43:15 -0400 Subject: [Freeswitch-users] Portaudio: Call For Input In-Reply-To: References: Message-ID: <20100601184315.GI20836@hijacked.us> On Sun, May 30, 2010 at 03:09:33PM -0400, Mitch Capper wrote: > Recently I have started attempting to replace an existing sip client with > freeswitch. In doing so I have started looking more towards portaudio. My > goal is to add some robustness to portaudio to make it better suited to > being used by sip clients. Some of the features I have currently worked > to implement: > > *Event generation for call holding and resuming > *Ability to switch input/output devices during calls with live audio going > on (useful for say speakerphone support) These are both good ideas. I've definitely wanted to switch the devices during a call and not been able to. > *Ability to switch both the indev and outdev at once rather than separately > (time savings) I'm not convinced this is really worth it, but it probably won't hurt anything. > *Ability to keep the audio stream initialized rather than each time a call > is made/ device is rang (time savings) > *Only init codecs during load / config reload rather than on audio stream > init (time savings) I think these should be done with a lot of caution. I believe the first one used to be the cause but it caused some weird issues..? > *Ability to call play (for sounds) with active calls going on Another good idea. > > Most of the improvements so far have been to make portaudio a bit faster at > doing things and less restricted than previously coded. Most changes have > been made through additional configuration variables so out of the box > portaudio will not function any differently than currently. My request for > input is on next steps and warnings. > > Right now portaudio keeps track of two streams, the call audio stream and > the ring audio stream (and both are inited on demand only accept with my > always active stream change). > > I was thinking of taking the always active streams to a slightly higher > level: > I am thinking about moving to have it keep track of an arbitrary number of > streams, a linked list of streams it is keeping track of with no more than > one stream per input/output pair. This would allow for initing a device > prior to use to allow for near instant use of that stream. This would > remove the small delay that still exists (under 1 second currently I would > estimate) when say switching on the speaker phone. Or allow for very quick > playing of audio on a specific device. > Aside from the speed increases in stream switching this would allow for you > to play audio on a device that isnt currently the primary audio device. > While I do not plan to take it this far currently, it would actually allow > for a much easier time of handling multiple calls on different devices at > the same time. > > The other way to go is to look towards trying to speed up initing streams > more, there are some yields in the code that look like safety things that > may be possible to remove without negative affects. The problem is, a lot of them are there for a reason. Dig into the commit logs (svn blame or git blame) and see *why* and *when* they were added. Don't assume that they're just there because of 'cargo cult programming', they were added for a reason. > > > Also I looked at the last merge of portaudio from upstream into trunk (end > of 2007) and the changes that were made back then. The good news is there > were not actually a lot of base code changes and most of them have actually > been merged into upstream now so updating to the latest may not be a very > hard thing to do. This is in part to see if it helps with the audio > quality issues that people of portaudio seem to report as it certainly is > not something you want in your client. I believe I was behind the last sync with upstream (fix for FreeBSD). If you want to try running it with a newer portaudio, by all means try it, but TEST IT THOROUGHLY - this means linux/mac/windows and maybe even solaris. Making it better on windows but worse everywhere else will not make you popular. > > If anyone has any input into the current state of portaudio or purposed > changes please let me know. In addition if anyone has input into getting > better call quality out of PA that would be extremely advantageous, as other > than updating to trunk I doubt my changes will result in much of a quality > improvement. I am specifically also interested if anyone knows of any of > the reasons some of the safeguards that are there are in place, some of > which I don't see a technical reason for them to be there unless there is > funnyness in libportaudio itself (which certainly could be and may be > partially resolved with updating to the changes from the last 3 years). If > anyone knows any stress testing etc that presented issues for PA previously > that would be helpful. I am working with PA in windows, do have a linux box > I will do some limited testing on, but overall if you just can test various > changes let me know as that will be helpful too as I believe removing some > of the safety steps may not present as an issue right away. I could also add > a "faster" option to portaudio config that would result in the optimizations > rather than just removing them completely. > I would definitely like to see some work on portaudio, but be aware that the module works as well as it does because we tested it in all sorts of crazy situations and make it work in the majority of them and that you don't want to add regressions just to pursue some mythical performance/quality goal. Just do some investigating on why things are the way they are before you get to hacking, so you know what to test for. Good luck, Andrew From lloyd.aloysius at gmail.com Tue Jun 1 12:05:35 2010 From: lloyd.aloysius at gmail.com (Aloysius Lloyd) Date: Tue, 1 Jun 2010 15:05:35 -0400 Subject: [Freeswitch-users] LUA - ** - Disconnect the call or stop call progress [ ringing or slient] and jump to the next statement Message-ID: Hi All, I would like to implement a feature code ** - Disconnect the leg B or stop call progress [ ringing or slient] and jump to the next statement. Here is the Lua code I am using to bridge two calls. How to implement the feature code like ** for this purpose. session:preAnswer(); digits = session:playAndGetDigits(10, 20, 3, 5000, "#", "enter-dest.wav", "invalid-digits.wav", "\\d+|\\*"); obSession = freeswitch.Session("sofia/gateway/voipms/"..digits,session) if obSession:ready() then obSession:execute("sched_hangup","+60 alloted_timeout"); freeswitch.bridge(session, obSession); end Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/e8326e4c/attachment.html From msc at freeswitch.org Tue Jun 1 12:20:29 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Jun 2010 12:20:29 -0700 Subject: [Freeswitch-users] OSTAG - Open Source Telephony Action Group Message-ID: Good news! The Open Source Telephony Action Group (OSTAG) is almost ready to become an official 501(c)3 non-profit organization. We need to raise $800 in order to file the necessary paperwork. We have set up a PayPal account where we can all send our donations: donations at ostag.org Please assist us in moving this important organization forward. OSTAG is dedicated to the advancement of open source telecommuncations software that improves the lives of people all over the world. This is a most worthy cause! Please help now and in the future. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/0dc7fb40/attachment.html From steve.d.ward at gmail.com Tue Jun 1 12:42:47 2010 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 1 Jun 2010 12:42:47 -0700 Subject: [Freeswitch-users] Single Park & Retrieve Extension Message-ID: Hello list, A useful feature (e.g. for an operator attendant console) is to have the ability to park and retrieve a caller with the use of a single button. The idea is to have a bank of BLF/speed-dial buttons (I've done this on Aastra 6757i and Polycom 650), where each button represents a holding slot (existing on the FreeSWITCH system) for callers. Each slot can only hold one caller. If a caller is in the slot, your BLF light is on. Press the button and the caller is retrieved (you are connected with the caller). Transfer another caller to that button (to that button's destination), and the parked caller is retrieved from the slot and both callers are connected; and the slot is now free again. If a slot does not hold a caller, the light is off. Transfer a caller to that "button", and the caller is held in the slot. I know I appreciate the very useful mod_fifo module and the help I've gotten from anthm on IRC in understanding the way FS does fifos. With that acquired understanding, I was able to implement the above feature with some simple fifo config and some concise dialplan. I just thought I'd share a little of this - especially since I'm under the impression there have been others interested in implementing something like this. Here's one way of going about it, which has proven to be exactly what was needed for a project I'm working on. If you're interested, here is a little page I've written up to describe it: http://wiki.freeswitch.org/wiki/Park_%26_Retrieve<%20http://wiki.freeswitch.org/wiki/Park_%26_Retrieve> (Now linked to from the mod_fifo page examples section as well.) - Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/a6c05c07/attachment.html From phone.bytes at gmail.com Tue Jun 1 14:42:15 2010 From: phone.bytes at gmail.com (Phone) Date: Tue, 01 Jun 2010 15:42:15 -0600 Subject: [Freeswitch-users] Questions on Building an application for FreeSWITCH In-Reply-To: References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com> <4BFBE6C2.5000703@gmail.com> <4BFBEF7C.9020107@gmail.com> <9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400> <4BFC4A64.5090208@gmail.com> <4BFD4562.5030804@gmail.com> <4BFEE96A.2010003@gmail.com> Message-ID: <4C057EB7.3090005@gmail.com> Thanks for the sample Bob! I am looking to do this in a linux environment...trying to get access to Visual Studio 2008 to explore your example. Anyone with examples that could be implemented on the linux side would be appreciated also. Thanks for all the comments and feedback...awesome forum! Bob Coleman wrote: > Ok > > On Mon, May 31, 2010 at 9:33 PM, Jan Berger wrote: > >> Bob, >> >> I suggest you write a wiki-article on FS about this, because this email is >> soon forgotten. >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob >> Coleman >> Sent: 30. mai 2010 23:56 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Questions on Building an application for >> FreeSWITCH >> >> I have put up an example for the outbound eventsocket in c#, it is >> real basic but gives you one approach to using this method. This has >> been coupled together by looking at other samples. >> >> There are two files, one is the dialplan ( 810_sample_ivr.xml ) which >> should be placed in the conf/dialplan/default folder >> The other file is a zip including a library for talking to the event >> socket and a simple test listener. >> >> http://www.devassert.com/apps/freeswitch/810_sample_ivr.xml >> http://www.devassert.com/apps/freeswitch/freeswitch.es.net.zip >> >> The password on the zip is cluecon >> >> To try the test, have the dialplan in place and run freeswitch, >> connect to an extension with a soft phone, eg 1001, dial 810 and >> listen. >> >> Assuming you have freeswitch callie prompts installed >> sounds\en\us\callie\conference\8000 all will be good. >> >> Sample is in Visual Studio 2008 and uses a threading library written >> by another party. >> >> If you have any questions just ask. >> >> On Fri, May 28, 2010 at 9:51 AM, Phone wrote: >> >>> Thanks, >>> >>> Examples and feedback are most helpful! >>> >>> Bob Coleman wrote: >>> >>>> Hi, >>>> >>>> Will incorporate some threading into the example I am working on for you. >>>> >>>> The outbound event socket method is very similar to the dialogic >>>> environment, I know how you felt though, but by starting small(like >>>> just even answering a call) gets you moving pretty quick. Freeswitch >>>> is lots of fun to work with, and the guys on here are very >>>> supportive!! >>>> >>>> Bob >>>> >>>> On Thu, May 27, 2010 at 3:59 AM, Phone wrote: >>>> >>>> >>>>> Thanks to all for the most helpful feedback. Sharing your approaches >>>>> and experiences are a big help. I look forward to the upcoming code >>>>> samples. >>>>> >>>>> I was coming from a windows/dialogic environment where I used a library >>>>> that allowed me to work on a little higher level. For example, I had a >>>>> call to "play a file" that took a parameter of whether or not to allow a >>>>> dtmf to interrupt. There was also a call to "ReadDtmfs" that took >>>>> parameters to specify the number of Dtmf's to read, how long to wait for >>>>> them, and what terminating character to use. I guess that you could >>>>> write some scripts or compiled code with these same types of functions >>>>> to simplify some of these routine tasks with reusable code? >>>>> >>>>> Also, the library handled the threading and scheduling with the OS. I >>>>> am still unclear on handling the events. I guess you have a big loop >>>>> reading events and then acting on them using the uuid to determine which >>>>> call it is and how to deal with the next step of the call? Any feedback >>>>> on this part of the project? >>>>> >>>>> Again, Thanks! >>>>> >>>>> Bob Coleman wrote: >>>>> >>>>> >>>>>> Ah sorry, I started with the esl to get an understanding then wrote my >>>>>> own socket library(was actually very easy to do), when I mean docs I >>>>>> mean the event socket docs. I still think of it as the esl, my >>>>>> mistake. >>>>>> >>>>>> http://wiki.freeswitch.org/wiki/Event_Socket >>>>>> >>>>>> I started with a codeplex project, that had been abandoned, and then >>>>>> once I understood the structure of the event socket language, was able >>>>>> to rewrite it to better handle what we were doing. >>>>>> >>>>>> I also married it up to an old gotdotnet asterisk fast agi project, >>>>>> once again abandoned, to allow for the use of asterisk as well, but in >>>>>> the end freeswitch won because we could use just one platform. >>>>>> >>>>>> I am busy writing a small sample app at the moment to demonstrate a >>>>>> problem I am trying to solve. Can release that code once sorted. Will >>>>>> be in a week or so. Am intending it as a quick way of testing event >>>>>> sockets, and trying various commands etc. before commiting to coding >>>>>> something. >>>>>> >>>>>> Bob >>>>>> >>>>>> On Wed, May 26, 2010 at 2:26 PM, Jan Berger >>>>>> >> wrote: >> >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> Do you have some sample code you could share + what docs did you look >>>>>>> >> at? >> >>>>>>> I would like to write and test some C# using ESL for my own work. >>>>>>> >>>>>>> Jan >>>>>>> >>>>>>> -----Original Message----- >>>>>>> From: freeswitch-users-bounces at lists.freeswitch.org >>>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>>>>>> >> Bob >> >>>>>>> Coleman >>>>>>> Sent: 26. mai 2010 04:07 >>>>>>> To: freeswitch-users at lists.freeswitch.org >>>>>>> Subject: Re: [Freeswitch-users] Questions on Building an application >>>>>>> >> for >> >>>>>>> FreeSWITCH >>>>>>> >>>>>>> We used c# as the rest of our systems are windows based. The language >>>>>>> doesnt matter too much, as long as you know where you are headed, what >>>>>>> performance you require, and what platform you are going to be using. >>>>>>> >>>>>>> Found the ESL so much easier than the dialogic c library we were >>>>>>> >> using. >> >>>>>>> The docs for the esl are easy to understand, the thing I couldnt get >>>>>>> my head around initially was the dialing out, with the dialogic you >>>>>>> are in the middle when you dial, ie already on the channel, but with >>>>>>> freeswitch you are kind of the third party when you dial, the channel >>>>>>> being created by the dialing and handing it off to be worked on. We >>>>>>> make the call via an inbound event socket and hand it off to an >>>>>>> outbound event socket application via the dialplan. >>>>>>> >>>>>>> On Wed, May 26, 2010 at 10:08 AM, Phone wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>>> Thanks for the info. What language did you use? >>>>>>>> >>>>>>>> Bob Coleman wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>> Have just recently completed a project to convert an old windows >>>>>>>>> dialogic application(vb6) to FreeSWITCH, would reccommend using the >>>>>>>>> ESL, was able to map the old dialogic calls to the ESL calls pretty >>>>>>>>> easily. We used a mixture of inbound and outbound sockets, as we >>>>>>>>> >> have >> >>>>>>>>> people dialing us, not just dialing out etc. >>>>>>>>> >>>>>>>>> With the dialogic you open a port and make the call and handle the >>>>>>>>> dtmf, with freeswitch you create a socket connection to FreeSWITCH >>>>>>>>> >> to >> >>>>>>>>> dial the number and then hand it off to an extension for processing >>>>>>>>> the dtmf(that is one approach any way) >>>>>>>>> >>>>>>>>> Bob >>>>>>>>> >>>>>>>>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins >>>>>>>>> >> >> >>>>>>>>> >>>>>>> wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>>>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger >>>>>>>>>> >> >> >>>>>>>>>> >>>>>>> wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>>>>>> Actually - before you get "to smart" - may I suggest that you >>>>>>>>>>> >> start >> >>>>>>>>>>> writing >>>>>>>>>>> - or improving - the getting started sections of the doc. Address >>>>>>>>>>> >> the >> >>>>>>>>>>> areas >>>>>>>>>>> where you struggle and let others benefit from your work. >>>>>>>>>>> >>>>>>>>>>> I have been through similar issues myself - FS is one of the >>>>>>>>>>> >> easier >> >>>>>>>>>>> projects >>>>>>>>>>> to work with once you get under the hood, but you basically need >>>>>>>>>>> >> to >> >>>>>>>>>>> >>>>>>> evolve >>>>>>> >>>>>>> >>>>>>> >>>>>>>>>>> to the level where you read the source code. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> And if you can wait 2+ months for "the book" then that should help >>>>>>>>>> >> as >> >>>>>>>>>> >>>>>>> well. >>>>>>> >>>>>>> >>>>>>> >>>>>>>>>> :D >>>>>>>>>> -MC >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> >>>>>>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>>>>>>> http://www.freeswitch.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 >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> >>>>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>>>>> http://www.freeswitch.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 >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >>>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>>>> http://www.freeswitch.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 >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.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 >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > From mitch.capper at gmail.com Tue Jun 1 15:22:42 2010 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 1 Jun 2010 18:22:42 -0400 Subject: [Freeswitch-users] Portaudio: Call For Input In-Reply-To: <20100601184315.GI20836@hijacked.us> References: <20100601184315.GI20836@hijacked.us> Message-ID: Thanks, I will definitely review the full commit history on the portaudio files. The regression is my worst fear, I don't have the boxes, test cases, or time to really test the aggressive changeset as best as properly should be. I think I will concentrate on ensuring as minimal as possible changes out of the box and add configuration options to enable some of the optimizations which should allow users looking for the boosts or additional functionality to enable it without the possible negative affects to existing users. Syncing with upstream however is something that probably should be aggressively tested and there isn't a good way to just make that optional. The good news is mod_portaudio and libportaudio are pretty separate so I believe that at worst there will be a patchset to bring libportaudio up to trunk if people desire to do that but trunk could remain out of date. Thanks for the input! ~Mitch PS. the reason for switching both indev and outdev at once was for speakerphone (where you switch both the input and output at the same time sequentially during an active call), originally I was doing blocking calls and the delay there was a bit annoying, seeing how most of the time taken was on opening the stream itself (which opens both input and output at once) having an api command to set them both effectively halved that time. On Tue, Jun 1, 2010 at 2:43 PM, Andrew Thompson wrote: > On Sun, May 30, 2010 at 03:09:33PM -0400, Mitch Capper wrote: > > Recently I have started attempting to replace an existing sip client with > > freeswitch. In doing so I have started looking more towards portaudio. > My > > goal is to add some robustness to portaudio to make it better suited to > > being used by sip clients. Some of the features I have currently > worked > > to implement: > > > > *Event generation for call holding and resuming > > *Ability to switch input/output devices during calls with live audio > going > > on (useful for say speakerphone support) > > These are both good ideas. I've definitely wanted to switch the devices > during a call and not been able to. > > > *Ability to switch both the indev and outdev at once rather than > separately > > (time savings) > > I'm not convinced this is really worth it, but it probably won't hurt > anything. > > > *Ability to keep the audio stream initialized rather than each time a > call > > is made/ device is rang (time savings) > > *Only init codecs during load / config reload rather than on audio stream > > init (time savings) > > I think these should be done with a lot of caution. I believe the first > one used to be the cause but it caused some weird issues..? > > > *Ability to call play (for sounds) with active calls going on > > Another good idea. > > > > Most of the improvements so far have been to make portaudio a bit faster > at > > doing things and less restricted than previously coded. Most changes > have > > been made through additional configuration variables so out of the box > > portaudio will not function any differently than currently. My request > for > > input is on next steps and warnings. > > > > Right now portaudio keeps track of two streams, the call audio stream and > > the ring audio stream (and both are inited on demand only accept with my > > always active stream change). > > > > I was thinking of taking the always active streams to a slightly higher > > level: > > I am thinking about moving to have it keep track of an arbitrary number > of > > streams, a linked list of streams it is keeping track of with no more > than > > one stream per input/output pair. This would allow for initing a > device > > prior to use to allow for near instant use of that stream. This would > > remove the small delay that still exists (under 1 second currently I > would > > estimate) when say switching on the speaker phone. Or allow for very > quick > > playing of audio on a specific device. > > Aside from the speed increases in stream switching this would allow for > you > > to play audio on a device that isnt currently the primary audio device. > > While I do not plan to take it this far currently, it would actually > allow > > for a much easier time of handling multiple calls on different devices at > > the same time. > > > > The other way to go is to look towards trying to speed up initing streams > > more, there are some yields in the code that look like safety things that > > may be possible to remove without negative affects. > > The problem is, a lot of them are there for a reason. Dig into the > commit logs (svn blame or git blame) and see *why* and *when* they were > added. Don't assume that they're just there because of 'cargo cult > programming', they were added for a reason. > > > > > > Also I looked at the last merge of portaudio from upstream into trunk > (end > > of 2007) and the changes that were made back then. The good news is > there > > were not actually a lot of base code changes and most of them have > actually > > been merged into upstream now so updating to the latest may not be a very > > hard thing to do. This is in part to see if it helps with the audio > > quality issues that people of portaudio seem to report as it certainly is > > not something you want in your client. > > I believe I was behind the last sync with upstream (fix for FreeBSD). If > you want to try running it with a newer portaudio, by all means try it, > but TEST IT THOROUGHLY - this means linux/mac/windows and maybe even > solaris. Making it better on windows but worse everywhere else will not > make you popular. > > > > > If anyone has any input into the current state of portaudio or purposed > > changes please let me know. In addition if anyone has input into > getting > > better call quality out of PA that would be extremely advantageous, as > other > > than updating to trunk I doubt my changes will result in much of a > quality > > improvement. I am specifically also interested if anyone knows of any > of > > the reasons some of the safeguards that are there are in place, some of > > which I don't see a technical reason for them to be there unless there is > > funnyness in libportaudio itself (which certainly could be and may be > > partially resolved with updating to the changes from the last 3 years). > If > > anyone knows any stress testing etc that presented issues for PA > previously > > that would be helpful. I am working with PA in windows, do have a linux > box > > I will do some limited testing on, but overall if you just can test > various > > changes let me know as that will be helpful too as I believe removing > some > > of the safety steps may not present as an issue right away. I could also > add > > a "faster" option to portaudio config that would result in the > optimizations > > rather than just removing them completely. > > > > I would definitely like to see some work on portaudio, but be aware that > the module works as well as it does because we tested it in all sorts of > crazy situations and make it work in the majority of them and that you > don't want to add regressions just to pursue some mythical > performance/quality goal. > > Just do some investigating on why things are the way they are before you > get to hacking, so you know what to test for. > > Good luck, > > Andrew > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/6d649ea7/attachment-0001.html From bobc at devassert.com Tue Jun 1 15:27:33 2010 From: bobc at devassert.com (Bob Coleman) Date: Wed, 2 Jun 2010 10:27:33 +1200 Subject: [Freeswitch-users] Questions on Building an application for FreeSWITCH In-Reply-To: <4C057EB7.3090005@gmail.com> References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com> <4BFBE6C2.5000703@gmail.com> <4BFBEF7C.9020107@gmail.com> <9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400> <4BFC4A64.5090208@gmail.com> <4BFD4562.5030804@gmail.com> <4BFEE96A.2010003@gmail.com> <4C057EB7.3090005@gmail.com> Message-ID: Ok, should work fine in the express version(free) On Wed, Jun 2, 2010 at 9:42 AM, Phone wrote: > Thanks for the sample Bob! ?I am looking to do this in a linux > environment...trying to get access to Visual Studio 2008 to explore your > example. > > Anyone with examples that could be implemented on the linux side would > be appreciated also. > > Thanks for all the comments and feedback...awesome forum! > > Bob Coleman wrote: >> Ok >> >> On Mon, May 31, 2010 at 9:33 PM, Jan Berger wrote: >> >>> Bob, >>> >>> I suggest you write a wiki-article on FS about this, because this email is >>> soon forgotten. >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob >>> Coleman >>> Sent: 30. mai 2010 23:56 >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Questions on Building an application for >>> FreeSWITCH >>> >>> I have put up an example for the outbound eventsocket in c#, it is >>> real basic but gives you one approach to using this method. This has >>> been coupled together by looking at other samples. >>> >>> There are two files, one is the dialplan ( 810_sample_ivr.xml ) which >>> should be placed in the conf/dialplan/default folder >>> The other file is a zip including a library for talking to the event >>> socket and a simple test listener. >>> >>> http://www.devassert.com/apps/freeswitch/810_sample_ivr.xml >>> http://www.devassert.com/apps/freeswitch/freeswitch.es.net.zip >>> >>> The password on the zip is cluecon >>> >>> To try the test, have the dialplan in place and run freeswitch, >>> connect to an extension with a soft phone, eg 1001, dial 810 and >>> listen. >>> >>> Assuming you have freeswitch callie prompts installed >>> sounds\en\us\callie\conference\8000 all will be good. >>> >>> Sample is in Visual Studio 2008 and uses a threading library written >>> by another party. >>> >>> If you have any questions just ask. >>> >>> On Fri, May 28, 2010 at 9:51 AM, Phone wrote: >>> >>>> Thanks, >>>> >>>> Examples and feedback are most helpful! >>>> >>>> Bob Coleman wrote: >>>> >>>>> Hi, >>>>> >>>>> Will incorporate some threading into the example I am working on for you. >>>>> >>>>> The outbound event socket method is very similar to the dialogic >>>>> environment, I know how you felt though, but by starting small(like >>>>> just even answering a call) gets you moving pretty quick. Freeswitch >>>>> is lots of fun to work with, and the guys on here are very >>>>> supportive!! >>>>> >>>>> Bob >>>>> >>>>> On Thu, May 27, 2010 at 3:59 AM, Phone wrote: >>>>> >>>>> >>>>>> Thanks to all for the most helpful feedback. ?Sharing your approaches >>>>>> and experiences are a big help. ?I look forward to the upcoming code >>>>>> samples. >>>>>> >>>>>> I was coming from a windows/dialogic environment where I used a library >>>>>> that allowed me to work on a little higher level. ?For example, I had a >>>>>> call to "play a file" that took a parameter of whether or not to allow a >>>>>> dtmf to interrupt. ?There was also a call to "ReadDtmfs" that took >>>>>> parameters to specify the number of Dtmf's to read, how long to wait for >>>>>> them, and what terminating character to use. ?I guess that you could >>>>>> write some scripts or compiled code with these same types of functions >>>>>> to simplify some of these routine tasks with reusable code? >>>>>> >>>>>> Also, the library handled the threading and scheduling with the OS. ?I >>>>>> am still unclear on handling the events. ?I guess you have a big loop >>>>>> reading events and then acting on them using the uuid to determine which >>>>>> call it is and how to deal with the next step of the call? ?Any feedback >>>>>> on this part of the project? >>>>>> >>>>>> Again, Thanks! >>>>>> >>>>>> Bob Coleman wrote: >>>>>> >>>>>> >>>>>>> Ah sorry, I started with the esl to get an understanding then wrote my >>>>>>> own socket library(was actually very easy to do), when I mean docs I >>>>>>> mean the event socket docs. I still think of it as the esl, my >>>>>>> mistake. >>>>>>> >>>>>>> http://wiki.freeswitch.org/wiki/Event_Socket >>>>>>> >>>>>>> I started with a codeplex project, that had been abandoned, and then >>>>>>> once I understood the structure of the event socket language, was able >>>>>>> to rewrite it to better handle what we were doing. >>>>>>> >>>>>>> I also married it up to an old gotdotnet asterisk fast agi project, >>>>>>> once again abandoned, to allow for the use of asterisk as well, but in >>>>>>> the end freeswitch won because we could use just one platform. >>>>>>> >>>>>>> I am busy writing a small sample app at the moment to demonstrate a >>>>>>> problem I am trying to solve. Can release that code once sorted. Will >>>>>>> be in a week or so. Am intending it as a quick way of testing event >>>>>>> sockets, and trying various commands etc. before commiting to coding >>>>>>> something. >>>>>>> >>>>>>> Bob >>>>>>> >>>>>>> On Wed, May 26, 2010 at 2:26 PM, Jan Berger >>>>>>> >>> wrote: >>> >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> Do you have some sample code you could share + what docs did you look >>>>>>>> >>> at? >>> >>>>>>>> I would like to write and test some C# using ESL for my own work. >>>>>>>> >>>>>>>> Jan >>>>>>>> >>>>>>>> -----Original Message----- >>>>>>>> From: freeswitch-users-bounces at lists.freeswitch.org >>>>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>>>>>>> >>> Bob >>> >>>>>>>> Coleman >>>>>>>> Sent: 26. mai 2010 04:07 >>>>>>>> To: freeswitch-users at lists.freeswitch.org >>>>>>>> Subject: Re: [Freeswitch-users] Questions on Building an application >>>>>>>> >>> for >>> >>>>>>>> FreeSWITCH >>>>>>>> >>>>>>>> We used c# as the rest of our systems are windows based. The language >>>>>>>> doesnt matter too much, as long as you know where you are headed, what >>>>>>>> performance you require, and what platform you are going to be using. >>>>>>>> >>>>>>>> Found the ESL so much easier than the dialogic c library we were >>>>>>>> >>> using. >>> >>>>>>>> The docs for the esl are easy to understand, the thing I couldnt get >>>>>>>> my head around initially was the dialing out, with the dialogic you >>>>>>>> are in the middle when you dial, ie already on the channel, but with >>>>>>>> freeswitch you are kind of the third party when you dial, the channel >>>>>>>> being created by the dialing and handing it off to be worked on. We >>>>>>>> make the call via an inbound event socket and hand it off to an >>>>>>>> outbound event socket application via the dialplan. >>>>>>>> >>>>>>>> On Wed, May 26, 2010 at 10:08 AM, Phone wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>> Thanks for the info. ?What language did you use? >>>>>>>>> >>>>>>>>> Bob Coleman wrote: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>> Have just recently completed a project to convert an old windows >>>>>>>>>> dialogic application(vb6) to FreeSWITCH, would reccommend using the >>>>>>>>>> ESL, was able to map the old dialogic calls to the ESL calls pretty >>>>>>>>>> easily. We used a mixture of inbound and outbound sockets, as we >>>>>>>>>> >>> have >>> >>>>>>>>>> people dialing us, not just dialing out etc. >>>>>>>>>> >>>>>>>>>> With the dialogic you open a port and make the call and handle the >>>>>>>>>> dtmf, with freeswitch you create a socket connection to FreeSWITCH >>>>>>>>>> >>> to >>> >>>>>>>>>> dial the number and then hand it off to an extension for processing >>>>>>>>>> the dtmf(that is one approach any way) >>>>>>>>>> >>>>>>>>>> Bob >>>>>>>>>> >>>>>>>>>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins >>>>>>>>>> >>> >>> >>>>>>>>>> >>>>>>>> wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger >>>>>>>>>>> >>> >>> >>>>>>>>>>> >>>>>>>> wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>>>>> Actually - before you get "to smart" - may I suggest that you >>>>>>>>>>>> >>> start >>> >>>>>>>>>>>> writing >>>>>>>>>>>> - or improving - the getting started sections of the doc. Address >>>>>>>>>>>> >>> the >>> >>>>>>>>>>>> areas >>>>>>>>>>>> where you struggle and let others benefit from your work. >>>>>>>>>>>> >>>>>>>>>>>> I have been through similar issues myself - FS is one of the >>>>>>>>>>>> >>> easier >>> >>>>>>>>>>>> projects >>>>>>>>>>>> to work with once you get under the hood, but you basically need >>>>>>>>>>>> >>> to >>> >>>>>>>>>>>> >>>>>>>> evolve >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>>>>> to the level where you read the source code. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> And if you can wait 2+ months for "the book" then that should help >>>>>>>>>>> >>> as >>> >>>>>>>>>>> >>>>>>>> well. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>>>> :D >>>>>>>>>>> -MC >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> >>>>>>>>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>>>>>>>>> http://www.freeswitch.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 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> >>>>>>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>>>>>>> http://www.freeswitch.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 >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> >>>>>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>>>>>> http://www.freeswitch.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 >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.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 >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 jmesquita at freeswitch.org Tue Jun 1 15:42:26 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 1 Jun 2010 19:42:26 -0300 Subject: [Freeswitch-users] Portaudio: Call For Input In-Reply-To: References: <20100601184315.GI20836@hijacked.us> Message-ID: I can try to help as much as I can with the testing, specially because that's what I use on FSComm, so... I would love to add to the wishlist integrating SpeexDSP's AEC, VAD, and other features available directly out of the PA stream so that it is more efficient. I failed to use what is in the core, maybe due to latency? Thanks, Jo?o Mesquita On Tue, Jun 1, 2010 at 7:22 PM, Mitch Capper wrote: > Thanks, I will definitely review the full commit history on the portaudio > files. The regression is my worst fear, I don't have the boxes, test cases, > or time to really test the aggressive changeset as best as properly should > be. I think I will concentrate on ensuring as minimal as possible changes > out of the box and add configuration options to enable some of the > optimizations which should allow users looking for the boosts or additional > functionality to enable it without the possible negative affects to existing > users. Syncing with upstream however is something that probably should be > aggressively tested and there isn't a good way to just make that optional. > The good news is mod_portaudio and libportaudio are pretty separate so I > believe that at worst there will be a patchset to bring libportaudio up to > trunk if people desire to do that but trunk could remain out of date. > > Thanks for the input! > > ~Mitch > PS. the reason for switching both indev and outdev at once was for > speakerphone (where you switch both the input and output at the same time > sequentially during an active call), originally I was doing blocking calls > and the delay there was a bit annoying, seeing how most of the time taken > was on opening the stream itself (which opens both input and output at once) > having an api command to set them both effectively halved that time. > > > On Tue, Jun 1, 2010 at 2:43 PM, Andrew Thompson wrote: > >> On Sun, May 30, 2010 at 03:09:33PM -0400, Mitch Capper wrote: >> > Recently I have started attempting to replace an existing sip client >> with >> > freeswitch. In doing so I have started looking more towards portaudio. >> My >> > goal is to add some robustness to portaudio to make it better suited to >> > being used by sip clients. Some of the features I have currently >> worked >> > to implement: >> > >> > *Event generation for call holding and resuming >> > *Ability to switch input/output devices during calls with live audio >> going >> > on (useful for say speakerphone support) >> >> These are both good ideas. I've definitely wanted to switch the devices >> during a call and not been able to. >> >> > *Ability to switch both the indev and outdev at once rather than >> separately >> > (time savings) >> >> I'm not convinced this is really worth it, but it probably won't hurt >> anything. >> >> > *Ability to keep the audio stream initialized rather than each time a >> call >> > is made/ device is rang (time savings) >> > *Only init codecs during load / config reload rather than on audio >> stream >> > init (time savings) >> >> I think these should be done with a lot of caution. I believe the first >> one used to be the cause but it caused some weird issues..? >> >> > *Ability to call play (for sounds) with active calls going on >> >> Another good idea. >> > >> > Most of the improvements so far have been to make portaudio a bit faster >> at >> > doing things and less restricted than previously coded. Most changes >> have >> > been made through additional configuration variables so out of the box >> > portaudio will not function any differently than currently. My request >> for >> > input is on next steps and warnings. >> > >> > Right now portaudio keeps track of two streams, the call audio stream >> and >> > the ring audio stream (and both are inited on demand only accept with my >> > always active stream change). >> > >> > I was thinking of taking the always active streams to a slightly higher >> > level: >> > I am thinking about moving to have it keep track of an arbitrary >> number of >> > streams, a linked list of streams it is keeping track of with no more >> than >> > one stream per input/output pair. This would allow for initing a >> device >> > prior to use to allow for near instant use of that stream. This would >> > remove the small delay that still exists (under 1 second currently I >> would >> > estimate) when say switching on the speaker phone. Or allow for very >> quick >> > playing of audio on a specific device. >> > Aside from the speed increases in stream switching this would allow for >> you >> > to play audio on a device that isnt currently the primary audio device. >> > While I do not plan to take it this far currently, it would actually >> allow >> > for a much easier time of handling multiple calls on different devices >> at >> > the same time. >> > >> > The other way to go is to look towards trying to speed up initing >> streams >> > more, there are some yields in the code that look like safety things >> that >> > may be possible to remove without negative affects. >> >> The problem is, a lot of them are there for a reason. Dig into the >> commit logs (svn blame or git blame) and see *why* and *when* they were >> added. Don't assume that they're just there because of 'cargo cult >> programming', they were added for a reason. >> > >> > >> > Also I looked at the last merge of portaudio from upstream into trunk >> (end >> > of 2007) and the changes that were made back then. The good news is >> there >> > were not actually a lot of base code changes and most of them have >> actually >> > been merged into upstream now so updating to the latest may not be a >> very >> > hard thing to do. This is in part to see if it helps with the audio >> > quality issues that people of portaudio seem to report as it certainly >> is >> > not something you want in your client. >> >> I believe I was behind the last sync with upstream (fix for FreeBSD). If >> you want to try running it with a newer portaudio, by all means try it, >> but TEST IT THOROUGHLY - this means linux/mac/windows and maybe even >> solaris. Making it better on windows but worse everywhere else will not >> make you popular. >> >> > >> > If anyone has any input into the current state of portaudio or purposed >> > changes please let me know. In addition if anyone has input into >> getting >> > better call quality out of PA that would be extremely advantageous, as >> other >> > than updating to trunk I doubt my changes will result in much of a >> quality >> > improvement. I am specifically also interested if anyone knows of any >> of >> > the reasons some of the safeguards that are there are in place, some of >> > which I don't see a technical reason for them to be there unless there >> is >> > funnyness in libportaudio itself (which certainly could be and may be >> > partially resolved with updating to the changes from the last 3 years). >> If >> > anyone knows any stress testing etc that presented issues for PA >> previously >> > that would be helpful. I am working with PA in windows, do have a linux >> box >> > I will do some limited testing on, but overall if you just can test >> various >> > changes let me know as that will be helpful too as I believe removing >> some >> > of the safety steps may not present as an issue right away. I could also >> add >> > a "faster" option to portaudio config that would result in the >> optimizations >> > rather than just removing them completely. >> > >> >> I would definitely like to see some work on portaudio, but be aware that >> the module works as well as it does because we tested it in all sorts of >> crazy situations and make it work in the majority of them and that you >> don't want to add regressions just to pursue some mythical >> performance/quality goal. >> >> Just do some investigating on why things are the way they are before you >> get to hacking, so you know what to test for. >> >> Good luck, >> >> Andrew >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100601/5c8c6365/attachment-0001.html From brian at freeswitch.org Tue Jun 1 15:48:05 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 1 Jun 2010 17:48:05 -0500 Subject: [Freeswitch-users] Questions on Building an application for FreeSWITCH In-Reply-To: References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com> <4BFBE6C2.5000703@gmail.com> <4BFBEF7C.9020107@gmail.com> <9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400> <4BFC4A64.5090208@gmail.com> <4BFD4562.5030804@gmail.com> <4BFEE96A.2010003@gmail.com> <4C057EB7.3090005@gmail.com> Message-ID: <50037FB5-DE1A-485C-980D-8F6ECAC3928E@freeswitch.org> Moving forward can we please highlight the area of the email we wish to reply to and then click reply, include the relevant portions of the email we are replying to... I see no need to have the past 40 emails in the reply on the thread. It wastes time, bandwidth and disk space and is harder to follow when trying to see if its something I need to pay attention to. /b On Jun 1, 2010, at 5:27 PM, Bob Coleman wrote: > Ok, should work fine in the express version(free) From andrew at hijacked.us Tue Jun 1 15:50:34 2010 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 1 Jun 2010 18:50:34 -0400 Subject: [Freeswitch-users] Portaudio: Call For Input In-Reply-To: References: <20100601184315.GI20836@hijacked.us> Message-ID: <20100601225034.GJ20836@hijacked.us> On Tue, Jun 01, 2010 at 06:22:42PM -0400, Mitch Capper wrote: > Thanks, I will definitely review the full commit history on the portaudio > files. The regression is my worst fear, I don't have the boxes, test cases, > or time to really test the aggressive changeset as best as properly should > be. I think I will concentrate on ensuring as minimal as possible changes > out of the box and add configuration options to enable some of the > optimizations which should allow users looking for the boosts or additional > functionality to enable it without the possible negative affects to existing > users. Syncing with upstream however is something that probably should be > aggressively tested and there isn't a good way to just make that optional. > The good news is mod_portaudio and libportaudio are pretty separate so I > believe that at worst there will be a patchset to bring libportaudio up to > trunk if people desire to do that but trunk could remain out of date. If bringing portaudio up to date breaks anything we SHOULD fix that anyway, unless they decided to redo the whole API or something. It does look like there's some worthwhile changes in there (maybe even dmix support for ALSA). The only thing that looks scary is that it seems they're using scons for building now, instead of make. If you manage to get the module working against trunk (and trunk building in-tree) that's probably a big step forwards. > > Thanks for the input! > > ~Mitch > PS. the reason for switching both indev and outdev at once was for > speakerphone (where you switch both the input and output at the same time > sequentially during an active call), originally I was doing blocking calls > and the delay there was a bit annoying, seeing how most of the time taken > was on opening the stream itself (which opens both input and output at once) > having an api command to set them both effectively halved that time. > That's fair, just make the API reasonable. Andrew From neilp at cs.stanford.edu Tue Jun 1 16:38:58 2010 From: neilp at cs.stanford.edu (Neil Patel) Date: Tue, 1 Jun 2010 16:38:58 -0700 Subject: [Freeswitch-users] does session:read flush digits? Message-ID: Before lua's session:read returns, does it do session:flushDigits so that the next call won't have the input carried over? The reason I ask is because it seems that after session:streamFile, any dtmf input hangs around for a subsequent session:read call, unless I explicitly invoke session:flushDigits after the streamFile call. Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/fd482c53/attachment.html From ranjtech at gmail.com Tue Jun 1 18:53:49 2010 From: ranjtech at gmail.com (RR) Date: Tue, 1 Jun 2010 21:53:49 -0400 Subject: [Freeswitch-users] Direct inward dialling In-Reply-To: <35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org> References: <491B305B.4060307@gmx.net> <072A175C-9004-4C14-90EC-9A93A8453787@gmail.com> <35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org> Message-ID: Hi Michael, I have another question for you and I REALLY REALLY apologise for abusing you as a Regex teacher but I have spent 2 hrs on this and can't figure out why this isn't working and believe me I have tried SEVERAL combinations but I can't make this work. the question is, why does the regex ^\+?1?(0[0-1]+)?([32|48|54|55|65]\d+)\;?(phone-context=)?\+?(\d+)?$ match 414xxxxxxxx;phone-context=+41 The intention is to ONLY match numbers that may be +32xxxxxxxxxxxxx, 032xxxxxxxxxxx,01132xxxxxxxxxxxxxx,+01132xxxxxxxxxxxx +48xxxxxxxxxxxxx, 048xxxxxxxxxxx,01148xxxxxxxxxxxxxx,+01148xxxxxxxxxxxx +54xxxxxxxxxxxxx, 054xxxxxxxxxxx,01154xxxxxxxxxxxxxx,+01154xxxxxxxxxxxx etc. why's it matching a number starting with 41? Is this incorrect RegEx implementation in FS or do I not get Regex? Sorry, I'll understand if you don't want to respond or help :) Thanks RR On Wed, May 26, 2010 at 1:40 AM, Michael S Collins wrote: > It's all good. Now you have to pay it forward. :) > -MC > > Sent from my iPhone > > On May 25, 2010, at 9:34 PM, RR wrote: > > Michael, > > Thank you SO SO much for the help. Your regex work perfectly as desired. I > had tried what you suggested earlier but I think I might've made a mistake > somewhere because I wasn't getting the right results so I resorted to doing > the "|" between the prefixes to strip them out thinking maybe FS works by > going if it begins + OR +1 OR 011 then remove them but I guess it doesn't as > when "\" appears it uses / matches against only the first one of the those > as opposed to all of those. > > Thanks again and sorry for wasting your time ;) > > Cheers > RR > > On Tue, May 25, 2010 at 9:19 PM, Michael Collins < > msc at freeswitch.org> wrote: > >> >> >> On Tue, May 25, 2010 at 5:44 PM, RR < >> ranjtech at gmail.com> wrote: >> >>> Michael, haha, yeah they indeed are. That's why I'm routing based on $2, >>> but I still see the 1 and/or the 011 going through to the "bridge" >>> application. Why?? >>> >> Because your regex is wrong. :) It took me a while to figure it out. I'm >> surprised it worked at all. All the stuff you have inside the first set of >> parens is not behaving the way you think it should be. If I read your >> intentions correctly you're trying to strip off leading: >> + >> OR >> +1 >> OR >> 1 >> >> In the first regex. Correct? If ANI is NANPA-ish then try this in your >> first regex: >> ^\+?1?([2-9]\d+).*$ >> >> That should strip off leading + and/or 1 and capture just the 10-digit >> phone number in $1. (Be sure to use $1 and not $2, unless you had your heart >> set on using $2 in which case wrap the first part of the regex in parens) >> >> The other regex is also tricky. I assume you are trying to strip off the >> same as above as well as 011? Try this: >> ^\+?1?(011)?([2-9]\d+).*$ >> >> Again, if the phone number in question is NANPA then $2 should contain >> just the 10 digits you want. Play around with that and let us know what >> happens. Also, don't forget what I said about using regex from the fs_cli. >> You can test all this stuff yourself. :) >> >> -MC >> >> >> >> >>> >>> On Tue, May 25, 2010 at 8:34 PM, Michael Collins < >>> msc at freeswitch.org> wrote: >>> >>>> >>>> >>>> On Tue, May 25, 2010 at 5:27 PM, RR < >>>> ranjtech at gmail.com> wrote: >>>> >>>>> Ok, so I take that back. This seems to only work when the dialplan has >>>>> a specific ANI and DNIS / destination_number / sip_to_user defined. If this >>>>> is more general >>>>> >>>>> like >>>>> >>>>> >>>>> >>>>> >>>> break="never"> >>>>> >>>> data="effective_caller_id_number=$2"/> >>>>> >>>>> >>>>> >>>> expression="^(\+1?|\+|1?|011?)(\d+).*$" break="never"> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/$2"/> >>>>> >>>>> >>>>> >>>>> >>>>> then even though the expression/conditions seem to match, none of the >>>>> digits are being stripped off. Shouldn't this be stripping off digits?? >>>>> >>>>> Here's the debug output: >>>>> >>>>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did] >>>>> ani(16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never >>>>> Dialplan: sofia/external/16469NNNNNN Action >>>>> set(effective_caller_id_number=16469NNNNNN) >>>>> Dialplan: sofia/external/16469NNNNNN Action >>>>> set(effective_caller_id_name=16469NNNNNN) >>>>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did] >>>>> ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~ >>>>> /^(\+1?|\+|1?|011?)(\d+).*$/ break=never >>>>> Dialplan: sofia/external/16469NNNNNN Action set(continue_on_fail=false) >>>>> Dialplan: sofia/external/16469NNNNNN Action >>>>> set(hangup_after_bridge=true) >>>>> Dialplan: sofia/external/16469NNNNNN Action >>>>> set(domain_name=208.72.186.166) >>>>> Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true) >>>>> Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades >>>>> 4200 !USER_BUSY) >>>>> Dialplan: sofia/external/16469NNNNNN Action >>>>> bridge({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/011390NNNNNNNNNN) >>>>> >>>>> why're the '1' in the ANI and '011' in the DNIS/sip_to_user being >>>>> stripped off??? >>>>> >>>> >>>> Regex 101 :) >>>> >>>> The 1 or the 011 are in $1 >>>> -MC >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/93508035/attachment-0001.html From ron.freeswitch at mcleodnet.com Tue Jun 1 19:57:46 2010 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Tue, 1 Jun 2010 19:57:46 -0700 Subject: [Freeswitch-users] Direct inward dialling In-Reply-To: References: <491B305B.4060307@gmx.net><072A175C-9004-4C14-90EC-9A93A8453787@gmail.com><35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org> Message-ID: Is [32|48|54|55|65] correct? Shouldn't it be (32|48|54|55|65) instead? ^\+?1?(0[0-1]+)?((32|48|54|55|65)\d+)\;?(phone-context=)?\+?(\d+)?$ Ron _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of RR Sent: Tuesday, June 01, 2010 6:54 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Direct inward dialling Hi Michael, I have another question for you and I REALLY REALLY apologise for abusing you as a Regex teacher but I have spent 2 hrs on this and can't figure out why this isn't working and believe me I have tried SEVERAL combinations but I can't make this work. the question is, why does the regex ^\+?1?(0[0-1]+)?([32|48|54|55|65]\d+)\;?(phone-context=)?\+?(\d+)?$ match 414xxxxxxxx;phone-context=+41 The intention is to ONLY match numbers that may be +32xxxxxxxxxxxxx, 032xxxxxxxxxxx,01132xxxxxxxxxxxxxx,+01132xxxxxxxxxxxx +48xxxxxxxxxxxxx, 048xxxxxxxxxxx,01148xxxxxxxxxxxxxx,+01148xxxxxxxxxxxx +54xxxxxxxxxxxxx, 054xxxxxxxxxxx,01154xxxxxxxxxxxxxx,+01154xxxxxxxxxxxx etc. why's it matching a number starting with 41? Is this incorrect RegEx implementation in FS or do I not get Regex? Sorry, I'll understand if you don't want to respond or help :) Thanks RR On Wed, May 26, 2010 at 1:40 AM, Michael S Collins wrote: It's all good. Now you have to pay it forward. :) -MC Sent from my iPhone On May 25, 2010, at 9:34 PM, RR wrote: Michael, Thank you SO SO much for the help. Your regex work perfectly as desired. I had tried what you suggested earlier but I think I might've made a mistake somewhere because I wasn't getting the right results so I resorted to doing the "|" between the prefixes to strip them out thinking maybe FS works by going if it begins + OR +1 OR 011 then remove them but I guess it doesn't as when "\" appears it uses / matches against only the first one of the those as opposed to all of those. Thanks again and sorry for wasting your time ;) Cheers RR On Tue, May 25, 2010 at 9:19 PM, Michael Collins < msc at freeswitch.org> wrote: On Tue, May 25, 2010 at 5:44 PM, RR < ranjtech at gmail.com> wrote: Michael, haha, yeah they indeed are. That's why I'm routing based on $2, but I still see the 1 and/or the 011 going through to the "bridge" application. Why?? Because your regex is wrong. :) It took me a while to figure it out. I'm surprised it worked at all. All the stuff you have inside the first set of parens is not behaving the way you think it should be. If I read your intentions correctly you're trying to strip off leading: + OR +1 OR 1 In the first regex. Correct? If ANI is NANPA-ish then try this in your first regex: ^\+?1?([2-9]\d+).*$ That should strip off leading + and/or 1 and capture just the 10-digit phone number in $1. (Be sure to use $1 and not $2, unless you had your heart set on using $2 in which case wrap the first part of the regex in parens) The other regex is also tricky. I assume you are trying to strip off the same as above as well as 011? Try this: ^\+?1?(011)?([2-9]\d+).*$ Again, if the phone number in question is NANPA then $2 should contain just the 10 digits you want. Play around with that and let us know what happens. Also, don't forget what I said about using regex from the fs_cli. You can test all this stuff yourself. :) -MC On Tue, May 25, 2010 at 8:34 PM, Michael Collins < msc at freeswitch.org> wrote: On Tue, May 25, 2010 at 5:27 PM, RR < ranjtech at gmail.com> wrote: Ok, so I take that back. This seems to only work when the dialplan has a specific ANI and DNIS / destination_number / sip_to_user defined. If this is more general like then even though the expression/conditions seem to match, none of the digits are being stripped off. Shouldn't this be stripping off digits?? Here's the debug output: Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did] ani(16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never Dialplan: sofia/external/16469NNNNNN Action set(effective_caller_id_number=16469NNNNNN) Dialplan: sofia/external/16469NNNNNN Action set(effective_caller_id_name=16469NNNNNN) Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did] ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~ /^(\+1?|\+|1?|011?)(\d+).*$/ break=never Dialplan: sofia/external/16469NNNNNN Action set(continue_on_fail=false) Dialplan: sofia/external/16469NNNNNN Action set(hangup_after_bridge=true) Dialplan: sofia/external/16469NNNNNN Action set(domain_name=208.72.186.166) Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true) Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades 4200 !USER_BUSY) Dialplan: sofia/external/16469NNNNNN Action bridge({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_bl ades)}/011390NNNNNNNNNN) why're the '1' in the ANI and '011' in the DNIS/sip_to_user being stripped off??? Regex 101 :) The 1 or the 011 are in $1 -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100601/6c54031d/attachment-0001.html From tculjaga at gmail.com Wed Jun 2 04:09:56 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 2 Jun 2010 13:09:56 +0200 Subject: [Freeswitch-users] time format In-Reply-To: <78B6C072-C1D4-4270-8AB0-C355984DD13D@gmail.com> References: <78B6C072-C1D4-4270-8AB0-C355984DD13D@gmail.com> Message-ID: On Tue, Jun 1, 2010 at 4:50 PM, David Ponzone wrote: > Tihomir, > > you can do that yourself with expr. > > h = expr floor(credit_time/3600) > m = expr floor(mod(credit_time,3600)/60) > s = expr mod(credit_time,60) > > Just convert that to the right $expr{} syntax. > > > cool, it works: Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/b400a67e/attachment.html From tculjaga at gmail.com Wed Jun 2 04:23:41 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 2 Jun 2010 13:23:41 +0200 Subject: [Freeswitch-users] time format In-Reply-To: References: <78B6C072-C1D4-4270-8AB0-C355984DD13D@gmail.com> Message-ID: On Wed, Jun 2, 2010 at 1:09 PM, Tihomir Culjaga wrote: > > > On Tue, Jun 1, 2010 at 4:50 PM, David Ponzone wrote: > >> Tihomir, >> >> you can do that yourself with expr. >> >> h = expr floor(credit_time/3600) >> m = expr floor(mod(credit_time,3600)/60) >> s = expr mod(credit_time,60) >> >> Just convert that to the right $expr{} syntax. >> >> >> > cool, it works: > > > > Thanks! > and finally its like this: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/61a8d62a/attachment.html From nagalenoj at gmail.com Wed Jun 2 05:18:51 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Wed, 2 Jun 2010 17:48:51 +0530 Subject: [Freeswitch-users] Channel variables difference Message-ID: Dear friends, I need to know the difference between these two values, Channel-Destination-Number and Caller-Destination-Number. I've found these two in the CHANNEL_DATA event header in event socket(when FS connects to an "Event Socket Outbound" handler). can someone help me please? -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/c227cd96/attachment.html From damjan at ecntelecoms.com Wed Jun 2 06:53:28 2010 From: damjan at ecntelecoms.com (Damjan Jovanovic) Date: Wed, 02 Jun 2010 15:53:28 +0200 Subject: [Freeswitch-users] How to do a "bridging" conference Message-ID: <1275486808.2055.73.camel@damjan-laptop> Hi I am looking for a way to have a single conference distributed over multiple Freeswitch servers. For example callers 1, 2 and 3 use a conference on Freeswitch server A, and callers 4, 5 and 6 use a conference on Freeswitch server B. These servers mix the local audio individually and only share a single audio channel between them. Is this what a "bridging" conference does? If not, what is it for? Also how could I efficiently make a large listen-only "conference" where there is only ever 1 talker, with very many listeners distributed over lots of servers? Thank you Damjan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/5caaba26/attachment.html From mcampbellsmith at gmail.com Wed Jun 2 07:14:36 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 2 Jun 2010 16:14:36 +0200 Subject: [Freeswitch-users] rtpmap question PCMU/PCMA In-Reply-To: References: <0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com> <088D4AC5-BE1D-4304-8139-FB00CFC796F9@jerris.com> <4C03F504.5070209@coppice.org> Message-ID: Coming back to my original question: Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'? Is it possible to add the end of the SDP parameters by doing something like: Can this be done anywhere in the dialplan as long as its before the bridge command? On Mon, May 31, 2010 at 9:15 PM, Mark Campbell-Smith wrote: > Thanks Guys. > > I should have noted that this is not related to an FS fault at all. > When using FS in bypass media mode, the call is rejected by the > client. ?Without bypass media mode it works. > > I just know there is a huge sip knowledge on this mailing list, and > would be able to get my answers easily. > > I guess the android client sipdroid does not like the broken and/or > shiny new SDP format. > > Cheers > > On Mon, May 31, 2010 at 7:42 PM, Steve Underwood wrote: >> Hi, >> >> I think its a perfectly reasonable invite, including the shiny new >> capabilities stuff which should reach full RFC status shortly. As it is >> new, I think the jury is currently out on whether existing poorly >> implemented SIP packages will choke on it. >> >> Steve >> >> >> On 06/01/2010 12:35 AM, Michael Jerris wrote: >>> This seems to be a badly broken sdp attempting to offer audio and t.38 but missing the m=image line from the sdp. >>> >>> Mike >>> >>> On May 31, 2010, at 10:57 AM, Mark Campbell-Smith wrote: >>> >>> >>>> Hi David, >>>> >>>> Its an INVITE. ?Full invite below: >>>> >>>> ? ?INVITE sip:gw+Phonzo at 124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0 >>>> ? ?Record-Route: >>>> ? ?Via: SIP/2.0/UDP >>>> 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0 >>>> ? ?Via: SIP/2.0/UDP >>>> 80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061 >>>> ? ?Max-Forwards: 16 >>>> ? ?From: 010711xxxx >>>> ;tag=262787ae2a9104a0c7700794a69028aco >>>> ? ?To: >>>> ? ?Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE. >>>> ? ?CSeq: 200 INVITE >>>> ? ?Contact: Anonymous >>>> ? ?Expires: 300 >>>> ? ?User-Agent: Sippy >>>> ? ?cisco-GUID: 1658214937-1822691807-2631794736-96895450 >>>> ? ?h323-conf-id: 1658214937-1822691807-2631794736-96895450 >>>> ? ?Content-disposition: session >>>> ? ?Content-Length: 364 >>>> ? ?Content-Type: application/sdp >>>> >>>> ? ?v=0 >>>> ? ?o=Sippy 141730476 0 IN IP4 80.232.37.178 >>>> ? ?s=- >>>> ? ?t=0 0 >>>> ? ?m=audio 47676 RTP/AVP 8 0 18 101 >>>> ? ?c=IN IP4 213.50.91.3 >>>> ? ?a=fmtp:18 annexb=yes >>>> ? ?a=rtpmap:101 telephone-event/8000 >>>> ? ?a=fmtp:101 0-15 >>>> ? ?a=sqn: 0 >>>> ? ?a=cdsc: 1 audio RTP/AVP ?8 >>>> ? ?a=cdsc: 2 image udptl t38 >>>> ? ?a=cpar: a=T38FaxUdpEC:t38UDPRedundancy >>>> ? ?a=cpar: a=T38FaxVersion:0 >>>> ? ?a=cpar: a=T38MaxBitRate:14400 >>>> ? ?a=sendrecv >>>> >>>> Regards >>>> Mark >>>> >>>> On Mon, May 31, 2010 at 4:29 PM, David Ponzone ?wrote: >>>> >>>>> Mark, >>>>> This looks like a T38 Re-INVITE, but a weird one. >>>>> David Ponzone ?Direction Technique >>>>> email: david.ponzone at ipeva.fr >>>>> tel: ? ? ?01 74 03 18 97 >>>>> gsm: ? 06 66 98 76 34 >>>>> Service Client IPeva >>>>> tel: ? ? ?0811 46 26 26 >>>>> www.ipeva.fr ?- ? www.ipeva-studio.com >>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible >>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il >>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce >>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. >>>>> >>>>> >>>>> >>>>> Le 31/05/2010 ? 16:12, Mark Campbell-Smith a ?crit : >>>>> >>>>> Hi All, >>>>> >>>>> I'm sure I've discussed this before, but I searched through my gmail >>>>> and google and couldn't find the answer. >>>>> >>>>> Below is the SDP parameters from my sip provider. ?Is it mandatory to >>>>> always include the rtpmap details for PCMU/PCMA codes? ?For example >>>>> something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'? >>>>> >>>>> I'm using sipdroid on android and it rejects this with 'codec not supported' >>>>> >>>>> Thanks >>>>> >>>>> >>>>> >>>>> o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx >>>>> s=- >>>>> t=0 0 >>>>> m=audio 47676 RTP/AVP 8 0 18 101 >>>>> c=IN IP4 xxx.xxx.xxx.xxx >>>>> a=fmtp:18 annexb=yes >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-15 >>>>> a=sqn: 0 >>>>> a=cdsc: 1 audio RTP/AVP ?8 >>>>> a=cdsc: 2 image udptl t38 >>>>> a=cpar: a=T38FaxUdpEC:t38UDPRedundancy >>>>> a=cpar: a=T38FaxVersion:0 >>>>> a=cpar: a=T38MaxBitRate:14400 >>>>> ] >>>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> > From brian at freeswitch.org Wed Jun 2 07:21:54 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 09:21:54 -0500 Subject: [Freeswitch-users] rtpmap question PCMU/PCMA In-Reply-To: References: <0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com> <088D4AC5-BE1D-4304-8139-FB00CFC796F9@jerris.com> <4C03F504.5070209@coppice.org> Message-ID: <93EDBB96-A71E-48D4-9F4E-9AEF92C0DE81@freeswitch.org> No it is NOT mandatory to include them. Only when they are on defined codec numbers. /b On Jun 2, 2010, at 9:14 AM, Mark Campbell-Smith wrote: > Coming back to my original question: > Is it mandatory to always include the rtpmap details for PCMU/PCMA > codes? For example something like 'a=rtpmap:8 PCMA/8000' and > 'a=rtpmap:0 PCMU/8000'? > > Is it possible to add the end of the SDP parameters by doing something like: > > > data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/> > data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/> > > > > Can this be done anywhere in the dialplan as long as its before the > bridge command? From pray at theprossergroup.com Wed Jun 2 05:11:09 2010 From: pray at theprossergroup.com (Praveen Ray) Date: Wed, 2 Jun 2010 08:11:09 -0400 Subject: [Freeswitch-users] No Early Media Ringback Message-ID: Hi All I am using Freeswitch 1.0.6 with Vitelity as inbound/outbound provider. My Inbound calls route properly to extension 1000, however, the caller does not hear the extension ringing. The extension at 1000 does ring however and once I pick up the call, the conversation proceeds normally. I just can't figure out why the caller hears silence during phone ringing. I have tried putting instant_ringback in dialplan/public.xml: The log lines during inbound calls show: 2010-06-02 07:55:15.913827 [NOTICE] mod_sofia.c:1992 Pre-Answer sofia/external/7816409134 at 64.2.142.15! 2010-06-02 07:55:15.916855 [DEBUG] switch_core_session.c:703 Send signal sofia/internal/sip:1000 at 172.168.128.193:5060 [BREAK] 2010-06-02 07:55:15.916855 [DEBUG] sofia.c:4167 Channel sofia/external/ 2123135987 at 64.2.142.15 skipping state [early][183] 2010-06-02 07:55:15.916855 [DEBUG] switch_core_session.c:642 Send signal sofia/external/2123135987 at 64.2.142.15 [BREAK] 2010-06-02 07:55:15.916855 [DEBUG] switch_ivr_originate.c:1124 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2010-06-02 07:55:15.916855 [DEBUG] switch_core_codec.c:122 sofia/external/ 2123135987 at 64.2.142.15 Push codec L16:10 2010-06-02 07:55:15.916855 [DEBUG] switch_ivr_originate.c:1189 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2010-06-02 07:55:15.922793 [DEBUG] sofia.c:4172 Channel sofia/internal/ sip:1000 at 172.168.128.193:5060 entering state [calling][0] 2010-06-02 07:55:15.981765 [DEBUG] sofia.c:5866 IP 64.2.142.15 Rejected by acl "domains". Falling back to Digest auth. 2010-06-02 07:55:15.984797 [WARNING] sofia_reg.c:1873 Can't find user [ prav_24.90.81.72 at 172.168.128.203] You must define a domain called '172.168.128.203' in your directory and add a user with the id="prav_24.90.81.72" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2010-06-02 07:55:15.984797 [WARNING] sofia_reg.c:1033 SIP auth failure (INVITE) on sofia profile 'internal' for [5087203445 at 24.90.81.72] from ip 64.2.142.15 where 5087203445 is my inbound DID. Can someone please help decipher what the warning means and why there might not be any early media ringback.. thanks -Praveen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/63f61b4f/attachment-0001.html From gustavo.espeche at upper-soft.com Wed Jun 2 06:50:26 2010 From: gustavo.espeche at upper-soft.com (Gustavo Espeche) Date: Wed, 02 Jun 2010 10:50:26 -0300 Subject: [Freeswitch-users] xml_curl endpoint register Message-ID: <20100602105026.fkxak9i0pw0c0c4k@www.upper-soft.com> Hello all, i'm new in freeswitch we are migrating our software form yate to freeswitch. I'm writing a registration php script using xml_curl module, the registration was good but some endpoint don't use 5060 as sip port and when i register it, i can't find the endpoint port in curl post, because of then when i dial to an endpoint that isn't listen in sip port 5060 the call fail. Some one know how can i fix it? when i register the endpoint i insert in a mysql db time, ip, and when some one dial to this user i read my db for know the user's ip. but i need insert the endpoint sip's port too for it work. Best Regards. Gustavo Espeche www.easyipcall.com From brian at freeswitch.org Wed Jun 2 07:53:23 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 09:53:23 -0500 Subject: [Freeswitch-users] No Early Media Ringback In-Reply-To: References: Message-ID: <8D868F2A-2B7F-46DA-8D65-955A5C1C0282@freeswitch.org> Try setting the ringback variable to something you wish to use as ringback tone. /b On Jun 2, 2010, at 7:11 AM, Praveen Ray wrote: > I am using Freeswitch 1.0.6 with Vitelity as inbound/outbound provider. My Inbound calls route properly to extension 1000, however, the caller does not hear the extension ringing. The extension at 1000 does ring however and once I pick up the call, the conversation proceeds normally. I just can't figure out why the caller hears silence during phone ringing. I have tried putting instant_ringback in dialplan/public.xml: > From info at evestech.com Wed Jun 2 08:02:07 2010 From: info at evestech.com (Kashif Kahn) Date: Wed, 2 Jun 2010 08:02:07 -0700 (PDT) Subject: [Freeswitch-users] Robust Affordable Speech Recognition Message-ID: <269652.60932.qm@web203.biz.mail.re2.yahoo.com> Dear All, All those who have wanted a speech recognition solution for Freeswitch but found the software cost too expensive or the recognition accuracy unsatisfactory, please consider Vestec Speech Engine for Freeswitchat: http://www.vestec.ca/products A starter kit - which is a specially priced one port (ie. one channel) license for the standard engine - is available for only $25. Additional ports (channels) licenses can be purchased for $99/port. The engine comes with a free-of-charge Freeswitch connector, thereby allowing direct interaction via Dialplan. Best regards, -Kashif Kashif Kahn VP, Business Development Vestec, Inc. Waterloo, ON Canada phone: (519) 885-7615 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/0c47f844/attachment.html From mcampbellsmith at gmail.com Wed Jun 2 08:03:50 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 2 Jun 2010 17:03:50 +0200 Subject: [Freeswitch-users] rtpmap question PCMU/PCMA In-Reply-To: <93EDBB96-A71E-48D4-9F4E-9AEF92C0DE81@freeswitch.org> References: <0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com> <088D4AC5-BE1D-4304-8139-FB00CFC796F9@jerris.com> <4C03F504.5070209@coppice.org> <93EDBB96-A71E-48D4-9F4E-9AEF92C0DE81@freeswitch.org> Message-ID: Thanks Brian. I didnt think it was mandatory. As my client is not accepting this sdp format, can I change it? I've tried with and and and have late negotiation active, but nothing is changed in the SDP. Basically I want change v=0 o=Sippy 142967340 3687727620696226321 IN IP4 80.232.37.178 s=- t=0 0 m=audio 36824 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 to something like: v=0 o=Sippy 142967340 3687727620696226321 IN IP4 80.232.37.178 s=- t=0 0 m=audio 36824 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 Is there an 'easy' way to do this? Thanks On Wed, Jun 2, 2010 at 4:21 PM, Brian West wrote: > No it is NOT mandatory to include them. ?Only when they are on defined codec numbers. > > /b > > On Jun 2, 2010, at 9:14 AM, Mark Campbell-Smith wrote: > >> Coming back to my original question: >> Is it mandatory to always include the rtpmap details for PCMU/PCMA >> codes? ?For example something like 'a=rtpmap:8 PCMA/8000' and >> 'a=rtpmap:0 PCMU/8000'? >> >> Is it possible to add the end of the SDP parameters by doing something like: >> ? ? ? ? >> ? ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? ? ? ?> data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/> >> ? ? ? ? ? ? ? ? ? ? > data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/> >> ? ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? >> >> Can this be done anywhere in the dialplan as long as its before the >> bridge command? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 andrewkt at aktzero.com Wed Jun 2 08:17:09 2010 From: andrewkt at aktzero.com (Andrew Thompson) Date: Wed, 02 Jun 2010 11:17:09 -0400 Subject: [Freeswitch-users] No Early Media Ringback In-Reply-To: References: Message-ID: <4C0675F5.2050907@aktzero.com> On 6/2/2010 8:11 AM, Praveen Ray wrote: > Hi All > I am using Freeswitch 1.0.6 with Vitelity as inbound/outbound > provider. My Inbound calls route properly to extension 1000, however, > the caller does not hear the extension ringing. The extension at 1000 > does ring however and once I pick up the call, the conversation > proceeds normally. I just can't figure out why the caller hears > silence during phone ringing. I have tried putting instant_ringback in > dialplan/public.xml: I get this from a couple VOIP providers. My main number goes straight to a IVR so it's a little strange for some callers, but for the others ring_ready seems to help. If you immediately start doing stuff, like an IVR, it won't help unless you put in a delay(so the ringing can happen) before you answer. But the way I see it, once you've got the call, do something with it! If the inbound call just rings until someone answers, this should do it: -- Andrew Thompson From brian at freeswitch.org Wed Jun 2 08:25:09 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 10:25:09 -0500 Subject: [Freeswitch-users] No Early Media Ringback In-Reply-To: <4C0675F5.2050907@aktzero.com> References: <4C0675F5.2050907@aktzero.com> Message-ID: But in his case usage of instant_ringback would require the ringback variable to produce the ringing... the other issue is if you send a 180 some endpoints and devices might not generate local ringback. :P Welcome to SIP the land of the SHOULD and MAY words. /b On Jun 2, 2010, at 10:17 AM, Andrew Thompson wrote: > From mgende at gendesign.com Wed Jun 2 08:28:42 2010 From: mgende at gendesign.com (Michael Gende) Date: Wed, 2 Jun 2010 10:28:42 -0500 Subject: [Freeswitch-users] Out-Going Call Transfer Question Message-ID: Hello, I wonder if someone could direct me to some documentation (I'm sure it exists) on how to undertake something. We've been happily using FS for some time now. Our is set-up emulates a pretty standard key system. To the point of my post: I can, for incoming calls, easily and successfully transfer them to conference rooms and other registered extensions. However, if I call out to another party, I can only transfer the call to a conference room, not another extension. I suspect this has to do with FS not knowing - or me not correctly telling it how - to handle the "out going" case. I.e, which "end" of the call to transfer. Sometimes its handy to make a call and transfer the callee to some other extension. Any suggested reading? Thanks, Mike G. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/5c5a03c2/attachment.html From cesar.bermudez at gmail.com Wed Jun 2 08:35:59 2010 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Wed, 2 Jun 2010 12:35:59 -0300 Subject: [Freeswitch-users] Serbian DID number. Message-ID: Sorry list, i'know this its no the correct spot to ask this, but i'dont have more places to ask, i'need a serbian did, anyone have any place to buy one? Best regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/24843d8d/attachment.html From infos at madovsky.org Wed Jun 2 08:43:56 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 2 Jun 2010 11:43:56 -0400 Subject: [Freeswitch-users] Serbian DID number. References: Message-ID: <2B73E97F85294BD28D4E4C06950905B7@MOBILEE1705> maybe didx ----- Original Message ----- From: Cesar Bermudez To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, June 02, 2010 11:35 AM Subject: [Freeswitch-users] Serbian DID number. Sorry list, i'know this its no the correct spot to ask this, but i'dont have more places to ask, i'need a serbian did, anyone have any place to buy one? Best regards. ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/04afb131/attachment.html From ranjtech at gmail.com Wed Jun 2 08:52:13 2010 From: ranjtech at gmail.com (RR) Date: Wed, 2 Jun 2010 11:52:13 -0400 Subject: [Freeswitch-users] Direct inward dialling In-Reply-To: References: <491B305B.4060307@gmx.net> <072A175C-9004-4C14-90EC-9A93A8453787@gmail.com> <35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org> Message-ID: Hi Ron, Thanks so much for that...I think you may be right. I'm testing this and it seems to be working as expected but will do further testing and let you know. Thanks again, RR On Tue, Jun 1, 2010 at 10:57 PM, Ron McLeod wrote: > Is [32|48|54|55|65] correct? Shouldn?t it be (32|48|54|55|65) > instead? > > > > > > ^\+?1?(0[0-1]+)?((32|48|54|55|65)\d+)\;?(phone-context=)?\+?(\d+)?$ > > > > > > Ron > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *RR > *Sent:* Tuesday, June 01, 2010 6:54 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Direct inward dialling > > > > Hi Michael, > > > > I have another question for you and I REALLY REALLY apologise for abusing > you as a Regex teacher but I have spent 2 hrs on this and can't figure out > why this isn't working and believe me I have tried SEVERAL combinations but > I can't make this work. > > > > the question is, why does the regex > > > > ^\+?1?(0[0-1]+)?([32|48|54|55|65]\d+)\;?(phone-context=)?\+?(\d+)?$ > > > > match > > > > 414xxxxxxxx;phone-context=+41 > > > > The intention is to ONLY match numbers that may be > > > > +32xxxxxxxxxxxxx, 032xxxxxxxxxxx,01132xxxxxxxxxxxxxx,+01132xxxxxxxxxxxx > > +48xxxxxxxxxxxxx, 048xxxxxxxxxxx,01148xxxxxxxxxxxxxx,+01148xxxxxxxxxxxx > > +54xxxxxxxxxxxxx, 054xxxxxxxxxxx,01154xxxxxxxxxxxxxx,+01154xxxxxxxxxxxx > > etc. > > > > why's it matching a number starting with 41? > > > > Is this incorrect RegEx implementation in FS or do I not get Regex? > > > > Sorry, I'll understand if you don't want to respond or help :) > > > > Thanks > > RR > > > > > > > > > > On Wed, May 26, 2010 at 1:40 AM, Michael S Collins > wrote: > > It's all good. Now you have to pay it forward. :) > > -MC > > Sent from my iPhone > > > On May 25, 2010, at 9:34 PM, RR wrote: > > Michael, > > > > Thank you SO SO much for the help. Your regex work perfectly as desired. I > had tried what you suggested earlier but I think I might've made a mistake > somewhere because I wasn't getting the right results so I resorted to doing > the "|" between the prefixes to strip them out thinking maybe FS works by > going if it begins + OR +1 OR 011 then remove them but I guess it doesn't as > when "\" appears it uses / matches against only the first one of the those > as opposed to all of those. > > > > Thanks again and sorry for wasting your time ;) > > > > Cheers > > RR > > On Tue, May 25, 2010 at 9:19 PM, Michael Collins < > msc at freeswitch.org> wrote: > > > > On Tue, May 25, 2010 at 5:44 PM, RR < > ranjtech at gmail.com> wrote: > > Michael, haha, yeah they indeed are. That's why I'm routing based on $2, > but I still see the 1 and/or the 011 going through to the "bridge" > application. Why?? > > Because your regex is wrong. :) It took me a while to figure it out. I'm > surprised it worked at all. All the stuff you have inside the first set of > parens is not behaving the way you think it should be. If I read your > intentions correctly you're trying to strip off leading: > + > OR > +1 > OR > 1 > > In the first regex. Correct? If ANI is NANPA-ish then try this in your > first regex: > ^\+?1?([2-9]\d+).*$ > > That should strip off leading + and/or 1 and capture just the 10-digit > phone number in $1. (Be sure to use $1 and not $2, unless you had your heart > set on using $2 in which case wrap the first part of the regex in parens) > > The other regex is also tricky. I assume you are trying to strip off the > same as above as well as 011? Try this: > ^\+?1?(011)?([2-9]\d+).*$ > > Again, if the phone number in question is NANPA then $2 should contain just > the 10 digits you want. Play around with that and let us know what happens. > Also, don't forget what I said about using regex from the fs_cli. You can > test all this stuff yourself. :) > > -MC > > > > > > > On Tue, May 25, 2010 at 8:34 PM, Michael Collins < > msc at freeswitch.org> wrote: > > > > On Tue, May 25, 2010 at 5:27 PM, RR < > ranjtech at gmail.com> wrote: > > Ok, so I take that back. This seems to only work when the dialplan has a > specific ANI and DNIS / destination_number / sip_to_user defined. If this is > more general > > > > like > > > > > > > > break="never"> > > > > > > > > expression="^(\+1?|\+|1?|011?)(\d+).*$" break="never"> > > > > > > > > > > > > data="{sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/$2"/> > > > > > > > > > > then even though the expression/conditions seem to match, none of the > digits are being stripped off. Shouldn't this be stripping off digits?? > > > > Here's the debug output: > > > > Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did] > ani(16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never > > Dialplan: sofia/external/16469NNNNNN Action > set(effective_caller_id_number=16469NNNNNN) > > Dialplan: sofia/external/16469NNNNNN Action > set(effective_caller_id_name=16469NNNNNN) > > Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did] > ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~ > /^(\+1?|\+|1?|011?)(\d+).*$/ break=never > > Dialplan: sofia/external/16469NNNNNN Action set(continue_on_fail=false) > > Dialplan: sofia/external/16469NNNNNN Action set(hangup_after_bridge=true) > > Dialplan: sofia/external/16469NNNNNN Action set(domain_name=208.72.186.166) > > Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true) > > Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades 4200 > !USER_BUSY) > > Dialplan: sofia/external/16469NNNNNN Action > bridge({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/011390NNNNNNNNNN) > > > > why're the '1' in the ANI and '011' in the DNIS/sip_to_user being stripped > off??? > > > Regex 101 :) > > The 1 or the 011 are in $1 > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > > > _______________________________________________ > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100602/dbad1c62/attachment-0001.html From anthony.minessale at gmail.com Wed Jun 2 08:57:27 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Jun 2010 10:57:27 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back Message-ID: Some squatter has had freeswitch.com for years and it's time we had it back. We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party panel to resolve the dispute. They are using the domain to pose as a VoIP site (if you keep loading http://www.freeswitch.com/ you can see it in action) Don't let creeps like this misuse the internet! Use the paypal button on the site to donate to the cause. Everyone who donates will get their name up on our thank you page on our site. Put "freeswitch.com UDRP and your name as you would like to see it on the thank-you page" in the note on the paypal form. -- 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/20100602/336ee979/attachment.html From infos at madovsky.org Wed Jun 2 09:36:21 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 2 Jun 2010 12:36:21 -0400 Subject: [Freeswitch-users] Help get freeswitch.com back References: Message-ID: whois gives this : Domain: freeswitch.com Date Registered: 11/14/06 Date Modified: 05/21/10 Expiry Date: 11/14/10 DNS1: ns1.above.com DNS2: ns2.above.com Registrant Freeswitch Quebec Domain Administrator Casier Postale 2490 St-Niolcas, QC (CA) G7 A 4X5 Administrative Contact Freeswitch Quebec Domain Administrator Casier Postale 2490 St-Niolcas, QC (CA) G7 A 4X5 tddomainnames at gail.com +1.614-954-1905 Technical Contact Freeswitch Quebec Domain Administrator Casier Postale 2490 St-Niolcas, QC (CA) G7 A 4X5 tddomainnames at gail.com +1.614-954-1905 Registrar: Rebel.com I think it's not a legitimate company, unless if these guys are registered as company in Canada, but .COM is a US company priority, so it should not be too hard to resolve this situation. Maybe I can help you on this matter. Regards ----- Original Message ----- From: Anthony Minessale To: Freeswitch-users ; freeswitch-dev at lists.freeswitch.org Sent: Wednesday, June 02, 2010 11:57 AM Subject: [Freeswitch-users] Help get freeswitch.com back Some squatter has had freeswitch.com for years and it's time we had it back. We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party panel to resolve the dispute. They are using the domain to pose as a VoIP site (if you keep loading http://www.freeswitch.com/ you can see it in action) Don't let creeps like this misuse the internet! Use the paypal button on the site to donate to the cause. Everyone who donates will get their name up on our thank you page on our site. Put "freeswitch.com UDRP and your name as you would like to see it on the thank-you page" in the note on the paypal form. -- 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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/2de00dcc/attachment.html From anthony.minessale at gmail.com Wed Jun 2 09:53:00 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Jun 2010 11:53:00 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: Message-ID: Sadly, we have followed up on that contact info and have determined that it is all fake. On Wed, Jun 2, 2010 at 11:36 AM, Madovsky wrote: > whois gives this : > > Domain: freeswitch.com > > Date Registered: 11/14/06 > Date Modified: 05/21/10 > Expiry Date: 11/14/10 > DNS1: ns1.above.com > DNS2: ns2.above.com > > Registrant > > Freeswitch Quebec > Domain Administrator > Casier Postale 2490 > St-Niolcas, QC (CA) > G7 A 4X5 > > Administrative Contact > > Freeswitch Quebec > Domain Administrator > Casier Postale 2490 > St-Niolcas, QC (CA) > G7 A 4X5 > tddomainnames at gail.com > +1.614-954-1905 > > > Technical Contact > > Freeswitch Quebec > Domain Administrator > Casier Postale 2490 > St-Niolcas, QC (CA) > G7 A 4X5 > tddomainnames at gail.com > +1.614-954-1905 > > > Registrar: Rebel.com > > I think it's not a legitimate company, unless if these guys are registered > as company in Canada, > but .COM is a US company priority, so it should not be too hard to resolve > this situation. > Maybe I can help you on this matter. > > Regards > > > ----- Original Message ----- > *From:* Anthony Minessale > *To:* Freeswitch-users ; > freeswitch-dev at lists.freeswitch.org > *Sent:* Wednesday, June 02, 2010 11:57 AM > *Subject:* [Freeswitch-users] Help get freeswitch.com back > > Some squatter has had freeswitch.com for years and it's time we had it > back. > > We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party > panel to resolve the dispute. > They are using the domain to pose as a VoIP site (if you keep loading > http://www.freeswitch.com/ you can see it in action) > > Don't let creeps like this misuse the internet! Use the paypal button on > the site to donate to the cause. > > Everyone who donates will get their name up on our thank you page on our > site. > Put "freeswitch.com UDRP and your name as you would like to see it on the > thank-you page" in the note on the paypal form. > > > > -- > 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 > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > -- 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/20100602/c3cc46f0/attachment-0001.html From msc at freeswitch.org Wed Jun 2 09:55:49 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Jun 2010 09:55:49 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly! Kamailio being discussed today Message-ID: Come on down: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_02 Daniel Constantin-Mierla (aka "miconda" or Twitter and IRC) is talking about Kamailio. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/e456afc4/attachment.html From brian at freeswitch.org Wed Jun 2 09:55:46 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 11:55:46 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: Message-ID: All that info is false... and the email address isn't even valid. The people that own gail.com don't know anything about it. /b On Jun 2, 2010, at 11:36 AM, Madovsky wrote: > I think it's not a legitimate company, unless if these guys are registered as company in Canada, > but .COM is a US company priority, so it should not be too hard to resolve this situation. > Maybe I can help you on this matter. > > Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/da150e4a/attachment.html From 12ukwn at gmail.com Wed Jun 2 10:23:01 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 2 Jun 2010 19:23:01 +0200 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: Message-ID: <20100602192301.6e5710b3@anubis.defcon1> Le Wed, 2 Jun 2010 11:55:46 -0500, Brian West a ?crit : > All that info is false... and the email address isn't even valid. The > people that own gail.com don't know anything about it. ... > > I think it's not a legitimate company, unless if these guys are > > registered as company in Canada, but .COM is a US company priority, so > > it should not be too hard to resolve this situation. Maybe I can help > > you on this matter. so complain to the authority: they don't like very much fakd IDs. -- I wouldn't marry her with a ten foot pole. From brian at freeswitch.org Wed Jun 2 10:27:42 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 12:27:42 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <20100602192301.6e5710b3@anubis.defcon1> References: <20100602192301.6e5710b3@anubis.defcon1> Message-ID: To do that you have to pay to file a complaint and have someone handle the dispute for you. Its not cheap to do this the right way. /b On Jun 2, 2010, at 12:23 PM, Jean-Yves F. Barbier wrote: > so complain to the authority: they don't like very much fakd IDs. > > -- > I wouldn't marry her with a ten foot pole. From darren at aleph-com.net Wed Jun 2 10:51:42 2010 From: darren at aleph-com.net (Darren Wiebe) Date: Wed, 02 Jun 2010 11:51:42 -0600 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: <20100602192301.6e5710b3@anubis.defcon1> Message-ID: <4C069A2E.3070207@aleph-com.net> On 02/06/2010 11:27 AM, Brian West wrote: > To do that you have to pay to file a complaint and have someone handle the dispute for you. Its not cheap to do this the right way. > > /b > > On Jun 2, 2010, at 12:23 PM, Jean-Yves F. Barbier wrote: > > >> so complain to the authority: they don't like very much fakd IDs. >> >> -- >> I wouldn't marry her with a ten foot pole. >> Would the people that own that domain be able to create that email address for you? If I remember correctly if you have control of the administrative contacts you should be able to take control of a domain. Maybe I'm missing something here. Darren Wiebe darren at aleph-com.net From brian at freeswitch.org Wed Jun 2 11:05:39 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 13:05:39 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <4C069A2E.3070207@aleph-com.net> References: <20100602192301.6e5710b3@anubis.defcon1> <4C069A2E.3070207@aleph-com.net> Message-ID: That would open them up for Legal issues. I did raise that as an option but quickly ruled that out as it would be illegal. /b On Jun 2, 2010, at 12:51 PM, Darren Wiebe wrote: > Would the people that own that domain be able to create that email > address for you? If I remember correctly if you have control of the > administrative contacts you should be able to take control of a domain. > Maybe I'm missing something here. > > Darren Wiebe > darren at aleph-com.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/61db816c/attachment.html From mcampbellsmith at gmail.com Wed Jun 2 11:07:09 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 2 Jun 2010 20:07:09 +0200 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: <20100602192301.6e5710b3@anubis.defcon1> Message-ID: All the links on the www.freeswitch.com page point to a domain called ndparking.com. This guy owns a lot of domain names. http://whois.domaintools.com/ndparking.com gives details about who owns that There are USA phone numbers and yahoo email addresses. On Wed, Jun 2, 2010 at 7:27 PM, Brian West wrote: > To do that you have to pay to file a complaint and have someone handle the dispute for you. ?Its not cheap to do this the right way. > > /b > > On Jun 2, 2010, at 12:23 PM, Jean-Yves F. Barbier wrote: > >> so complain to the authority: they don't like very much fakd IDs. >> >> -- >> I wouldn't marry her with a ten foot pole. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 12ukwn at gmail.com Wed Jun 2 11:08:24 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 2 Jun 2010 20:08:24 +0200 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: <20100602192301.6e5710b3@anubis.defcon1> Message-ID: <20100602200824.0a0d7ea7@anubis.defcon1> Le Wed, 2 Jun 2010 12:27:42 -0500, Brian West a ?crit : > To do that you have to pay to file a complaint and have someone handle > the dispute for you. Its not cheap to do this the right way. what a wonderful us world...... -- Good news. Ten weeks from Friday will be a pretty good day. From msc at freeswitch.org Wed Jun 2 11:17:59 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Jun 2010 11:17:59 -0700 Subject: [Freeswitch-users] Direct inward dialling In-Reply-To: References: <491B305B.4060307@gmx.net> <072A175C-9004-4C14-90EC-9A93A8453787@gmail.com> <35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org> Message-ID: On Tue, Jun 1, 2010 at 7:57 PM, Ron McLeod wrote: > Is [32|48|54|55|65] correct? Shouldn?t it be (32|48|54|55|65) > instead? > > > > > > ^\+?1?(0[0-1]+)?((32|48|54|55|65)\d+)\;?(phone-context=)?\+?(\d+)?$ > Also, you really don't need the dash in [0-1], just do [01] which means "match a 0 or a 1" -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/83113aba/attachment.html From msc at freeswitch.org Wed Jun 2 11:40:17 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Jun 2010 11:40:17 -0700 Subject: [Freeswitch-users] xml_curl endpoint register In-Reply-To: <20100602105026.fkxak9i0pw0c0c4k@www.upper-soft.com> References: <20100602105026.fkxak9i0pw0c0c4k@www.upper-soft.com> Message-ID: Welcome to FreeSWITCH! I think you'll find a lot of helpful folks around here. Also, don't forget to join the IRC channel: #freeswitch on irc.freenode.net where you can discuss things in realtime. I'm having trouble visualizing what you are doing. Can you expand your description of what you are doing? You might want to put some logs into pastebin.freeswitch.org and link to those logs in this thread. Thanks, MC On Wed, Jun 2, 2010 at 6:50 AM, Gustavo Espeche < gustavo.espeche at upper-soft.com> wrote: > Hello all, i'm new in freeswitch we are migrating our software form > yate to freeswitch. > I'm writing a registration php script using xml_curl module, the > registration was good but some endpoint don't use 5060 as sip port and > when i register it, i can't find the endpoint port in curl post, > because of then when i dial to an endpoint that isn't listen in sip > port 5060 the call fail. > Some one know how can i fix it? > when i register the endpoint i insert in a mysql db time, ip, and when > some one dial to this user i read my db for know the user's ip. > but i need insert the endpoint sip's port too for it work. > > Best Regards. > > Gustavo Espeche > www.easyipcall.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/53d20fda/attachment-0001.html From msc at freeswitch.org Wed Jun 2 11:42:54 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Jun 2010 11:42:54 -0700 Subject: [Freeswitch-users] Out-Going Call Transfer Question In-Reply-To: References: Message-ID: Mike, Maybe you could pastebin a log of this behavior. Also, is this a plain FS install with the default dialplan? If not then include your relevant dialplan entries. Lastly, are you on the most recent hit HEAD version of FS? Thanks, MC On Wed, Jun 2, 2010 at 8:28 AM, Michael Gende wrote: > Hello, > > I wonder if someone could direct me to some documentation (I'm sure it > exists) on how to undertake something. > > We've been happily using FS for some time now. Our is set-up emulates a > pretty standard key system. > > To the point of my post: I can, for incoming calls, easily and successfully > transfer them to conference rooms and other registered extensions. > > However, if I call out to another party, I can only transfer the call to a > conference room, not another extension. > > I suspect this has to do with FS not knowing - or me not correctly telling > it how - to handle the "out going" case. I.e, which "end" of the call to > transfer. > > Sometimes its handy to make a call and transfer the callee to some other > extension. Any suggested reading? > > Thanks, > > Mike G. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/c1723f89/attachment.html From mcampbellsmith at gmail.com Wed Jun 2 11:44:48 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 2 Jun 2010 20:44:48 +0200 Subject: [Freeswitch-users] rtpmap question PCMU/PCMA In-Reply-To: References: <0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com> <088D4AC5-BE1D-4304-8139-FB00CFC796F9@jerris.com> <4C03F504.5070209@coppice.org> <93EDBB96-A71E-48D4-9F4E-9AEF92C0DE81@freeswitch.org> Message-ID: Ahha... I found why it wasn't working. I have media_bypass active and it appears this does not work when media_bypass is active. Is there any reason for this? Why can't FreeSWITCH modify the signalling and then let the users negotiate which codec to use? If I modify the codec list to be one that is not supported by the server, then its my fault. On Wed, Jun 2, 2010 at 5:03 PM, Mark Campbell-Smith wrote: > Thanks Brian. ?I didnt think it was mandatory. ?As my client is not > accepting this sdp format, can I change it? > > I've tried with > > and > > and > ? ? ? ? ? ? data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/> > ? ? ? ? ? ? data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/> > > and have late negotiation active, but nothing is changed in the SDP. > > Basically I want change > ? v=0 > ? o=Sippy 142967340 3687727620696226321 IN IP4 80.232.37.178 > ? s=- > ? t=0 0 > ? m=audio 36824 RTP/AVP 8 0 18 101 > ? c=IN IP4 213.50.91.3 > ? a=fmtp:18 annexb=yes > ? a=rtpmap:101 telephone-event/8000 > ? a=fmtp:101 0-15 > ? a=sqn: 0 > ? a=cdsc: 1 audio RTP/AVP ?8 > ? a=cdsc: 2 image udptl t38 > ? a=cpar: a=T38FaxUdpEC:t38UDPRedundancy > ? a=cpar: a=T38FaxVersion:0 > ? a=cpar: a=T38MaxBitRate:14400 > > to something like: > ? v=0 > ? o=Sippy 142967340 3687727620696226321 IN IP4 80.232.37.178 > ? s=- > ? t=0 0 > ? m=audio 36824 RTP/AVP 8 0 18 101 > ? c=IN IP4 213.50.91.3 > ? a=rtpmap:101 telephone-event/8000 > ? a=fmtp:101 0-15 > ? a=rtpmap:8 PCMA/8000 > ? a=rtpmap:0 PCMU/8000 > > Is there an 'easy' way to do this? > Thanks > > > On Wed, Jun 2, 2010 at 4:21 PM, Brian West wrote: >> No it is NOT mandatory to include them. ?Only when they are on defined codec numbers. >> >> /b >> >> On Jun 2, 2010, at 9:14 AM, Mark Campbell-Smith wrote: >> >>> Coming back to my original question: >>> Is it mandatory to always include the rtpmap details for PCMU/PCMA >>> codes? ?For example something like 'a=rtpmap:8 PCMA/8000' and >>> 'a=rtpmap:0 PCMU/8000'? >>> >>> Is it possible to add the end of the SDP parameters by doing something like: >>> ? ? ? ? >>> ? ? ? ? ? ? ? ? ? >>> ? ? ? ? ? ? ? ? ? ? ?>> data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/> >>> ? ? ? ? ? ? ? ? ? ? >> data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/> >>> ? ? ? ? ? ? ? ? ? >>> ? ? ? ? ? ? >>> >>> Can this be done anywhere in the dialplan as long as its before the >>> bridge command? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 brian at freeswitch.org Wed Jun 2 11:48:44 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 13:48:44 -0500 Subject: [Freeswitch-users] rtpmap question PCMU/PCMA In-Reply-To: References: <0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com> <088D4AC5-BE1D-4304-8139-FB00CFC796F9@jerris.com> <4C03F504.5070209@coppice.org> <93EDBB96-A71E-48D4-9F4E-9AEF92C0DE81@freeswitch.org> Message-ID: <5757E845-E6CE-49ED-82DC-91EA7C983B60@freeswitch.org> Because we are not a proxy. /b On Jun 2, 2010, at 1:44 PM, Mark Campbell-Smith wrote: > Why can't FreeSWITCH modify the signalling and then let the users > negotiate which codec to use? If I modify the codec list to be one > that is not supported by the server, then its my fault. From msc at freeswitch.org Wed Jun 2 11:51:40 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Jun 2010 11:51:40 -0700 Subject: [Freeswitch-users] Single Park & Retrieve Extension In-Reply-To: References: Message-ID: Thanks for donating and documenting! We appreciate it. -MC On Tue, Jun 1, 2010 at 12:42 PM, Steven Ward wrote: > Hello list, > > A useful feature (e.g. for an operator attendant console) is to have the > ability to park and retrieve a caller with the use of a single button. > > The idea is to have a bank of BLF/speed-dial buttons (I've done this on > Aastra 6757i and Polycom 650), where each button represents a holding slot > (existing on the FreeSWITCH system) for callers. Each slot can only hold > one caller. If a caller is in the slot, your BLF light is on. Press the > button and the caller is retrieved (you are connected with the caller). > Transfer another caller to that button (to that button's destination), and > the parked caller is retrieved from the slot and both callers are connected; > and the slot is now free again. If a slot does not hold a caller, the light > is off. Transfer a caller to that "button", and the caller is held in the > slot. > > I know I appreciate the very useful mod_fifo module and the help I've > gotten from anthm on IRC in understanding the way FS does fifos. With that > acquired understanding, I was able to implement the above feature with some > simple fifo config and some concise dialplan. > > I just thought I'd share a little of this - especially since I'm under the > impression there have been others interested in implementing something like > this. > > Here's one way of going about it, which has proven to be exactly what was > needed for a project I'm working on. > > If you're interested, here is a little page I've written up to describe it: > > http://wiki.freeswitch.org/wiki/Park_%26_Retrieve > > (Now linked to from the mod_fifo page examples section as well.) > > - Steve > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/55e38fba/attachment.html From delorenzodesign at gmail.com Wed Jun 2 11:55:37 2010 From: delorenzodesign at gmail.com (Michael De Lorenzo) Date: Wed, 2 Jun 2010 14:55:37 -0400 Subject: [Freeswitch-users] Using XLite/Softphone with Freeswitch Message-ID: I'm trying to configure XLite as a softphone to use with our Freeswitch installation. It seems I can register successfully with FS, but if I try to dial someone (not another Freeswitch user, I haven't tried another Freeswitch user because I don't have any others and don't need that functionality) I receive a busy signal and a 480 Temporarily Unavailable response. >From what I can tell in my SIP captures, the domain for the callee isn't getting properly set. I'm not sure where to set the domain (IP of my gateway) so the call can be completed. Note 0.0.0.0 is the IP address (I removed the actual IP) of my Freeswitch gateway and 1.1.1.1 is the IP address of the softphone/client. What setting am I missing? The "To" variable should be phonenumber at ip-of-my-gateway, the gateway I'm trying to use is the same one that works fine for outbound calls executed from the Freeswitch console. Can someone point me in the right direction? --------------------------------------------------------------------------------------------------------------------------------------- freeswitch at myserver-fs-dev-2> recv 824 bytes from udp/[1.1.1.1]:5418 at 18:49:56.846935: ------------------------------------------------------------------------ INVITE sip:9735551234 at 0.0.0.0 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5418 ;branch=z9hG4bK-d8754z-cb423158626fee66-1---d8754z-;rport Max-Forwards: 70 Contact: To: "9735551234"> From: "John Doe" >;tag=e61b231e Call-ID: YTJmOGEzZjY2ZWE3ZDVlNTc2ODZkYmEyYjMzN2ViOTA. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 263 v=0 o=- 2 2 IN IP4 1.1.1.1 s=CounterPath X-Lite 3.0 c=IN IP4 1.1.1.1 t=0 0 m=audio 7912 RTP/AVP 107 0 8 101 a=alt:1 1 : cyh417lX YyY5ABYF 10.100.95.226 5068 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 398 bytes to udp/[1.1.1.1]:5418 at 18:49:56.847674: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.1:5418 ;branch=z9hG4bK-d8754z-cb423158626fee66-1---d8754z-;rport=5418 From: "John Doe" >;tag=e61b231e To: "9735551234"> Call-ID: YTJmOGEzZjY2ZWE3ZDVlNTc2ODZkYmEyYjMzN2ViOTA. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Content-Length: 0 ------------------------------------------------------------------------ 2010-06-02 14:49:56.848100 [NOTICE] switch_channel.c:669 New Channel sofia/internal/9727289377 at 0.0.0.0 [a38cc944-6e77-11df-afe3-f7ba0651215e] 2010-06-02 14:49:56.853307 [INFO] mod_dialplan_xml.c:418 Processing John Doe->9735551234 in context public 2010-06-02 14:49:56.855781 [NOTICE] switch_core_state_machine.c:185 sofia/internal/9727289377 at 0.0.0.0 has executed the last dialplan instruction, hanging up. 2010-06-02 14:49:56.855781 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/9727289377 at 0.0.0.0 [CS_EXECUTE] [NORMAL_CLEARING] send 902 bytes to udp/[1.1.1.1]:5418 at 18:49:56.862822: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 1.1.1.1:5418 ;branch=z9hG4bK-d8754z-cb423158626fee66-1---d8754z-;rport=5418 From: "John Doe" >;tag=e61b231e To: "9735551234" >;tag=Z7SBNmNvryStm Call-ID: YTJmOGEzZjY2ZWE3ZDVlNTc2ODZkYmEyYjMzN2ViOTA. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 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 Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "9735551234" >;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ recv 349 bytes from udp/[1.1.1.1]:5418 at 18:49:56.866344: ------------------------------------------------------------------------ ACK sip:9735551234 at 0.0.0.0 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5418 ;branch=z9hG4bK-d8754z-cb423158626fee66-1---d8754z-;rport To: "9735551234" >;tag=Z7SBNmNvryStm From: "John Doe" >;tag=e61b231e Call-ID: YTJmOGEzZjY2ZWE3ZDVlNTc2ODZkYmEyYjMzN2ViOTA. CSeq: 1 ACK Content-Length: 0 ------------------------------------------------------------------------ 2010-06-02 14:49:56.890355 [NOTICE] switch_core_session.c:1183 Session 1 (sofia/internal/9727289377 at 0.0.0.0) Ended 2010-06-02 14:49:56.890355 [NOTICE] switch_core_session.c:1185 Close Channel sofia/internal/9727289377 at 0.0.0.0 [CS_DESTROY] --------------------------------------------------------------------------------------------------------------------------------------- On Wed, Jun 2, 2010 at 12:53 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. Help get freeswitch.com back (Anthony Minessale) > 2. Re: Help get freeswitch.com back (Madovsky) > 3. Re: Help get freeswitch.com back (Anthony Minessale) > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: Freeswitch-users , > freeswitch-dev at lists.freeswitch.org > Date: Wed, 2 Jun 2010 10:57:27 -0500 > Subject: [Freeswitch-users] Help get freeswitch.com back > Some squatter has had freeswitch.com for years and it's time we had it > back. > > We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party > panel to resolve the dispute. > They are using the domain to pose as a VoIP site (if you keep loading > http://www.freeswitch.com/ you can see it in action) > > Don't let creeps like this misuse the internet! Use the paypal button on > the site to donate to the cause. > > Everyone who donates will get their name up on our thank you page on our > site. > Put "freeswitch.com UDRP and your name as you would like to see it on the > thank-you page" in the note on the paypal form. > > > > -- > 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 > > > ---------- Forwarded message ---------- > From: "Madovsky" > To: > Date: Wed, 2 Jun 2010 12:36:21 -0400 > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > whois gives this : > > Domain: freeswitch.com > > Date Registered: 11/14/06 > Date Modified: 05/21/10 > Expiry Date: 11/14/10 > DNS1: ns1.above.com > DNS2: ns2.above.com > > Registrant > > Freeswitch Quebec > Domain Administrator > Casier Postale 2490 > St-Niolcas, QC (CA) > G7 A 4X5 > > Administrative Contact > > Freeswitch Quebec > Domain Administrator > Casier Postale 2490 > St-Niolcas, QC (CA) > G7 A 4X5 > tddomainnames at gail.com > +1.614-954-1905 > > > Technical Contact > > Freeswitch Quebec > Domain Administrator > Casier Postale 2490 > St-Niolcas, QC (CA) > G7 A 4X5 > tddomainnames at gail.com > +1.614-954-1905 > > > Registrar: Rebel.com > > I think it's not a legitimate company, unless if these guys are registered > as company in Canada, > but .COM is a US company priority, so it should not be too hard to resolve > this situation. > Maybe I can help you on this matter. > > Regards > > > ----- Original Message ----- > *From:* Anthony Minessale > *To:* Freeswitch-users ; > freeswitch-dev at lists.freeswitch.org > *Sent:* Wednesday, June 02, 2010 11:57 AM > *Subject:* [Freeswitch-users] Help get freeswitch.com back > > Some squatter has had freeswitch.com for years and it's time we had it > back. > > We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party > panel to resolve the dispute. > They are using the domain to pose as a VoIP site (if you keep loading > http://www.freeswitch.com/ you can see it in action) > > Don't let creeps like this misuse the internet! Use the paypal button on > the site to donate to the cause. > > Everyone who donates will get their name up on our thank you page on our > site. > Put "freeswitch.com UDRP and your name as you would like to see it on the > thank-you page" in the note on the paypal form. > > > > -- > 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 > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 2 Jun 2010 11:53:00 -0500 > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > Sadly, we have followed up on that contact info and have determined that it > is all fake. > > > On Wed, Jun 2, 2010 at 11:36 AM, Madovsky wrote: > >> whois gives this : >> >> Domain: freeswitch.com >> >> Date Registered: 11/14/06 >> Date Modified: 05/21/10 >> Expiry Date: 11/14/10 >> DNS1: ns1.above.com >> DNS2: ns2.above.com >> >> Registrant >> >> Freeswitch Quebec >> Domain Administrator >> Casier Postale 2490 >> St-Niolcas, QC (CA) >> G7 A 4X5 >> >> Administrative Contact >> >> Freeswitch Quebec >> Domain Administrator >> Casier Postale 2490 >> St-Niolcas, QC (CA) >> G7 A 4X5 >> tddomainnames at gail.com >> +1.614-954-1905 >> >> >> Technical Contact >> >> Freeswitch Quebec >> Domain Administrator >> Casier Postale 2490 >> St-Niolcas, QC (CA) >> G7 A 4X5 >> tddomainnames at gail.com >> +1.614-954-1905 >> >> >> Registrar: Rebel.com >> >> I think it's not a legitimate company, unless if these guys are registered >> as company in Canada, >> but .COM is a US company priority, so it should not be too hard to resolve >> this situation. >> Maybe I can help you on this matter. >> >> Regards >> >> >> ----- Original Message ----- >> *From:* Anthony Minessale >> *To:* Freeswitch-users ; >> freeswitch-dev at lists.freeswitch.org >> *Sent:* Wednesday, June 02, 2010 11:57 AM >> *Subject:* [Freeswitch-users] Help get freeswitch.com back >> >> Some squatter has had freeswitch.com for years and it's time we had it >> back. >> >> We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party >> panel to resolve the dispute. >> They are using the domain to pose as a VoIP site (if you keep loading >> http://www.freeswitch.com/ you can see it in action) >> >> Don't let creeps like this misuse the internet! Use the paypal button on >> the site to donate to the cause. >> >> Everyone who donates will get their name up on our thank you page on our >> site. >> Put "freeswitch.com UDRP and your name as you would like to see it on the >> thank-you page" in the note on the paypal form. >> >> >> >> -- >> 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 >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> >> > > > -- > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/14c12a47/attachment-0001.html From brian at freeswitch.org Wed Jun 2 12:02:30 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 14:02:30 -0500 Subject: [Freeswitch-users] Using XLite/Softphone with Freeswitch In-Reply-To: References: Message-ID: 2010-06-02 14:49:56.853307 [INFO] mod_dialplan_xml.c:418 Processing John Doe->9735551234 in context public 2010-06-02 14:49:56.855781 [NOTICE] switch_core_state_machine.c:185 sofia/internal/9727289377 at 0.0.0.0 has executed the last dialplan instruction, hanging up. Its not matching anything in your dialplan. Press F8 /b On Jun 2, 2010, at 1:55 PM, Michael De Lorenzo wrote: > I'm trying to configure XLite as a softphone to use with our Freeswitch installation. It seems I can register successfully with FS, but if I try to dial someone (not another Freeswitch user, I haven't tried another Freeswitch user because I don't have any others and don't need that functionality) I receive a busy signal and a 480 Temporarily Unavailable response. > > From what I can tell in my SIP captures, the domain for the callee isn't getting properly set. I'm not sure where to set the domain (IP of my gateway) so the call can be completed. Note 0.0.0.0 is the IP address (I removed the actual IP) of my Freeswitch gateway and 1.1.1.1 is the IP address of the softphone/client. > > What setting am I missing? The "To" variable should be phonenumber at ip-of-my-gateway, the gateway I'm trying to use is the same one that works fine for outbound calls executed from the Freeswitch console. > > Can someone point me in the right direction? From infos at madovsky.org Wed Jun 2 12:04:47 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 2 Jun 2010 15:04:47 -0400 Subject: [Freeswitch-users] Help get freeswitch.com back References: <20100602192301.6e5710b3@anubis.defcon1> <20100602200824.0a0d7ea7@anubis.defcon1> Message-ID: <4B58D4D54F5647A59999A83BB2A40CC5@MOBILEE1705> So why not send some beautiful network packets to these guys ? ----- Original Message ----- From: "Jean-Yves F. Barbier" <12ukwn at gmail.com> To: Sent: Wednesday, June 02, 2010 2:08 PM Subject: Re: [Freeswitch-users] Help get freeswitch.com back > Le Wed, 2 Jun 2010 12:27:42 -0500, > Brian West a ?crit : > >> To do that you have to pay to file a complaint and have someone handle >> the dispute for you. Its not cheap to do this the right way. > > what a wonderful us world...... > > -- > Good news. Ten weeks from Friday will be a pretty good day. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 infos at madovsky.org Wed Jun 2 12:08:06 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 2 Jun 2010 15:08:06 -0400 Subject: [Freeswitch-users] Help get freeswitch.com back References: <20100602192301.6e5710b3@anubis.defcon1> Message-ID: <891BEFD20CB841678F66A9EB11508C44@MOBILEE1705> ICANN asks real admin info when you register a domain. in case of fake personal info the domain name should be removed. ----- Original Message ----- From: "Brian West" To: Sent: Wednesday, June 02, 2010 1:27 PM Subject: Re: [Freeswitch-users] Help get freeswitch.com back > To do that you have to pay to file a complaint and have someone handle the > dispute for you. Its not cheap to do this the right way. > > /b > > On Jun 2, 2010, at 12:23 PM, Jean-Yves F. Barbier wrote: > >> so complain to the authority: they don't like very much fakd IDs. >> >> -- >> I wouldn't marry her with a ten foot pole. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brian at freeswitch.org Wed Jun 2 12:12:25 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 14:12:25 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <891BEFD20CB841678F66A9EB11508C44@MOBILEE1705> References: <20100602192301.6e5710b3@anubis.defcon1> <891BEFD20CB841678F66A9EB11508C44@MOBILEE1705> Message-ID: Still takes money to file all the paperwork. Are you willing to do the stack of paperwork for that? /b On Jun 2, 2010, at 2:08 PM, Madovsky wrote: > ICANN asks real admin info when you register a domain. > in case of fake personal info the domain name should be removed. > > > > ----- Original Message ----- > From: "Brian West" > To: > Sent: Wednesday, June 02, 2010 1:27 PM > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > > >> To do that you have to pay to file a complaint and have someone handle the >> dispute for you. Its not cheap to do this the right way. >> >> /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/4c2919f3/attachment.html From anthony.minessale at gmail.com Wed Jun 2 12:19:59 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Jun 2010 14:19:59 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <4B58D4D54F5647A59999A83BB2A40CC5@MOBILEE1705> References: <20100602192301.6e5710b3@anubis.defcon1> <20100602200824.0a0d7ea7@anubis.defcon1> <4B58D4D54F5647A59999A83BB2A40CC5@MOBILEE1705> Message-ID: The only way to get it back for sure is to file the dispute with ICANN. Like I said in the original email, they charge 1300 to have one person decide on it and 2600 to have 3 people. Maybe they will sell it to us for less knowing that they will lose once we file the dispute but I doubt it. On Wed, Jun 2, 2010 at 2:04 PM, Madovsky wrote: > So why not send some beautiful network packets to these guys ? > > ----- Original Message ----- > From: "Jean-Yves F. Barbier" <12ukwn at gmail.com> > To: > Sent: Wednesday, June 02, 2010 2:08 PM > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > > > > Le Wed, 2 Jun 2010 12:27:42 -0500, > > Brian West a ?crit : > > > >> To do that you have to pay to file a complaint and have someone handle > >> the dispute for you. Its not cheap to do this the right way. > > > > what a wonderful us world...... > > > > -- > > Good news. Ten weeks from Friday will be a pretty good day. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > -- 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/20100602/b10f6c1d/attachment.html From 12ukwn at gmail.com Wed Jun 2 12:23:05 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 2 Jun 2010 21:23:05 +0200 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <891BEFD20CB841678F66A9EB11508C44@MOBILEE1705> References: <20100602192301.6e5710b3@anubis.defcon1> <891BEFD20CB841678F66A9EB11508C44@MOBILEE1705> Message-ID: <20100602212305.4acbb5d4@anubis.defcon1> Le Wed, 2 Jun 2010 15:08:06 -0400, "Madovsky" a ?crit : > ICANN asks real admin info when you register a domain. > in case of fake personal info the domain name should be removed. Yeah, that's what I meant; especially if addresses are faked. -- Life is the childhood of our immortality. -- Goethe From infos at madovsky.org Wed Jun 2 12:36:19 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 2 Jun 2010 15:36:19 -0400 Subject: [Freeswitch-users] Help get freeswitch.com back References: <20100602192301.6e5710b3@anubis.defcon1><891BEFD20CB841678F66A9EB11508C44@MOBILEE1705> Message-ID: We talk about justice. usually the victim complaints and the criminal must pay... the contrary is called "mafia" ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, June 02, 2010 3:12 PM Subject: Re: [Freeswitch-users] Help get freeswitch.com back Still takes money to file all the paperwork. Are you willing to do the stack of paperwork for that? /b On Jun 2, 2010, at 2:08 PM, Madovsky wrote: ICANN asks real admin info when you register a domain. in case of fake personal info the domain name should be removed. ----- Original Message ----- From: "Brian West" To: Sent: Wednesday, June 02, 2010 1:27 PM Subject: Re: [Freeswitch-users] Help get freeswitch.com back To do that you have to pay to file a complaint and have someone handle the dispute for you. Its not cheap to do this the right way. /b ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/4e8548a2/attachment-0001.html From kris at kriskinc.com Wed Jun 2 12:40:44 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 2 Jun 2010 15:40:44 -0400 Subject: [Freeswitch-users] Status of absolute_codec_string Message-ID: Hello everyone, Either I am misunderstanding absolute_codec_string or there is something going on with it in trunk. I had a git checkout from a month ago that exhibited this same behavior so I updated about an hour ago. It's still there (or I'm still confused). In short, I have a call that comes in from a remote endpoint that will advertise PCMU and G729 (in that order). When I bridge the call I want to remove G729 capability from the outbound leg in some cases so FS doesn't have to transcode. In other cases (controlled from the dialplan), I want to place G729 as the first codec (again so FS doesn't have to transcode) but I can force G729. I've tried regex on ep_codec_string and I've tried regex on the SDP. Both seemed to work well but what didn't work was setting absolute_codec_string from either of these. Even when I try to set it manually without any fancy conditions it doesn't appear to work. The codec prefs from the SIP profile are presented to the remote endpoint. Full debug here: http://pastebin.freeswitch.org/13088 Any ideas? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From delorenzodesign at gmail.com Wed Jun 2 12:45:38 2010 From: delorenzodesign at gmail.com (Michael De Lorenzo) Date: Wed, 2 Jun 2010 15:45:38 -0400 Subject: [Freeswitch-users] Using XLite/Softphone with Freeswitch2 Message-ID: Ok, what do I need to add to the dialplan? I haven't changed it at all from the default. On Wed, Jun 2, 2010 at 3:36 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: Using XLite/Softphone with Freeswitch (Brian West) > 2. Re: Help get freeswitch.com back (Madovsky) > 3. Re: Help get freeswitch.com back (Madovsky) > 4. Re: Help get freeswitch.com back (Brian West) > 5. Re: Help get freeswitch.com back (Anthony Minessale) > 6. Re: Help get freeswitch.com back (Jean-Yves F. Barbier) > 7. Re: Help get freeswitch.com back (Madovsky) > > > ---------- Forwarded message ---------- > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 2 Jun 2010 14:02:30 -0500 > Subject: Re: [Freeswitch-users] Using XLite/Softphone with Freeswitch > 2010-06-02 14:49:56.853307 [INFO] mod_dialplan_xml.c:418 Processing John > Doe->9735551234 in context public > 2010-06-02 14:49:56.855781 [NOTICE] switch_core_state_machine.c:185 > sofia/internal/9727289377 at 0.0.0.0 has executed the last dialplan > instruction, hanging up. > > Its not matching anything in your dialplan. > > Press F8 > > /b > > On Jun 2, 2010, at 1:55 PM, Michael De Lorenzo wrote: > > > I'm trying to configure XLite as a softphone to use with our Freeswitch > installation. It seems I can register successfully with FS, but if I try to > dial someone (not another Freeswitch user, I haven't tried another > Freeswitch user because I don't have any others and don't need that > functionality) I receive a busy signal and a 480 Temporarily Unavailable > response. > > > > From what I can tell in my SIP captures, the domain for the callee isn't > getting properly set. I'm not sure where to set the domain (IP of my > gateway) so the call can be completed. Note 0.0.0.0 is the IP address (I > removed the actual IP) of my Freeswitch gateway and 1.1.1.1 is the IP > address of the softphone/client. > > > > What setting am I missing? The "To" variable should be > phonenumber at ip-of-my-gateway, the gateway I'm trying to use is the same > one that works fine for outbound calls executed from the Freeswitch console. > > > > Can someone point me in the right direction? > > > > > > ---------- Forwarded message ---------- > From: "Madovsky" > To: > Date: Wed, 2 Jun 2010 15:04:47 -0400 > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > So why not send some beautiful network packets to these guys ? > > ----- Original Message ----- From: "Jean-Yves F. Barbier" < > 12ukwn at gmail.com> > To: > Sent: Wednesday, June 02, 2010 2:08 PM > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > > > Le Wed, 2 Jun 2010 12:27:42 -0500, >> Brian West a ?crit : >> >> To do that you have to pay to file a complaint and have someone handle >>> the dispute for you. Its not cheap to do this the right way. >>> >> >> what a wonderful us world...... >> >> -- >> Good news. Ten weeks from Friday will be a pretty good day. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > > ---------- Forwarded message ---------- > From: "Madovsky" > To: > Date: Wed, 2 Jun 2010 15:08:06 -0400 > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > ICANN asks real admin info when you register a domain. > in case of fake personal info the domain name should be removed. > > > > ----- Original Message ----- From: "Brian West" > To: > Sent: Wednesday, June 02, 2010 1:27 PM > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > > > To do that you have to pay to file a complaint and have someone handle the >> dispute for you. Its not cheap to do this the right way. >> >> /b >> >> On Jun 2, 2010, at 12:23 PM, Jean-Yves F. Barbier wrote: >> >> so complain to the authority: they don't like very much fakd IDs. >>> >>> -- >>> I wouldn't marry her with a ten foot pole. >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > > ---------- Forwarded message ---------- > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 2 Jun 2010 14:12:25 -0500 > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > Still takes money to file all the paperwork. Are you willing to do the > stack of paperwork for that? > > > /b > > On Jun 2, 2010, at 2:08 PM, Madovsky wrote: > > ICANN asks real admin info when you register a domain. > in case of fake personal info the domain name should be removed. > > > > ----- Original Message ----- > From: "Brian West" > To: > Sent: Wednesday, June 02, 2010 1:27 PM > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > > > To do that you have to pay to file a complaint and have someone handle the > > > dispute for you. Its not cheap to do this the right way. > > > /b > > > > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 2 Jun 2010 14:19:59 -0500 > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > The only way to get it back for sure is to file the dispute with ICANN. > Like I said in the original email, they charge 1300 to have one person > decide on it and 2600 to have 3 people. > Maybe they will sell it to us for less knowing that they will lose once we > file the dispute but I doubt it. > > > On Wed, Jun 2, 2010 at 2:04 PM, Madovsky wrote: > >> So why not send some beautiful network packets to these guys ? >> >> ----- Original Message ----- >> From: "Jean-Yves F. Barbier" <12ukwn at gmail.com> >> To: >> Sent: Wednesday, June 02, 2010 2:08 PM >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> >> >> > Le Wed, 2 Jun 2010 12:27:42 -0500, >> > Brian West a ?crit : >> > >> >> To do that you have to pay to file a complaint and have someone handle >> >> the dispute for you. Its not cheap to do this the right way. >> > >> > what a wonderful us world...... >> > >> > -- >> > Good news. Ten weeks from Friday will be a pretty good day. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.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 >> > > > > -- > 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 > > > ---------- Forwarded message ---------- > From: "Jean-Yves F. Barbier" <12ukwn at gmail.com> > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 2 Jun 2010 21:23:05 +0200 > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > Le Wed, 2 Jun 2010 15:08:06 -0400, > "Madovsky" a ?crit : > > > ICANN asks real admin info when you register a domain. > > in case of fake personal info the domain name should be removed. > > Yeah, that's what I meant; especially if addresses are faked. > > -- > Life is the childhood of our immortality. > -- Goethe > > > > > ---------- Forwarded message ---------- > From: "Madovsky" > To: > Date: Wed, 2 Jun 2010 15:36:19 -0400 > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > We talk about justice. > usually the victim complaints and the criminal must pay... > the contrary is called "mafia" > > > > ----- Original Message ----- > *From:* Brian West > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Wednesday, June 02, 2010 3:12 PM > *Subject:* Re: [Freeswitch-users] Help get freeswitch.com back > > Still takes money to file all the paperwork. Are you willing to do the > stack of paperwork for that? > > > /b > > On Jun 2, 2010, at 2:08 PM, Madovsky wrote: > > ICANN asks real admin info when you register a domain. > in case of fake personal info the domain name should be removed. > > > > ----- Original Message ----- > From: "Brian West" > To: > Sent: Wednesday, June 02, 2010 1:27 PM > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > > > To do that you have to pay to file a complaint and have someone handle the > > > dispute for you. Its not cheap to do this the right way. > > > /b > > > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100602/58075a53/attachment-0001.html From brian at freeswitch.org Wed Jun 2 12:48:34 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 14:48:34 -0500 Subject: [Freeswitch-users] Status of absolute_codec_string In-Reply-To: References: Message-ID: <584D7DCB-37BA-4E95-9CEF-719012B6CEFF@freeswitch.org> You can enable "late-negotiation", then set inherit_codec=true then it will delay the negotiation till it gets the answer from the B-Leg before the A-Leg's codec is thus accomplishing the same thing. /b On Jun 2, 2010, at 2:40 PM, Kristian Kielhofner wrote: > Hello everyone, > > Either I am misunderstanding absolute_codec_string or there is > something going on with it in trunk. I had a git checkout from a > month ago that exhibited this same behavior so I updated about an hour > ago. It's still there (or I'm still confused). > > In short, I have a call that comes in from a remote endpoint that > will advertise PCMU and G729 (in that order). When I bridge the call > I want to remove G729 capability from the outbound leg in some cases > so FS doesn't have to transcode. In other cases (controlled from the > dialplan), I want to place G729 as the first codec (again so FS > doesn't have to transcode) but I can force G729. I've tried regex on > ep_codec_string and I've tried regex on the SDP. Both seemed to work > well but what didn't work was setting absolute_codec_string from > either of these. > > Even when I try to set it manually without any fancy conditions it > doesn't appear to work. The codec prefs from the SIP profile are > presented to the remote endpoint. Full debug here: > > http://pastebin.freeswitch.org/13088 > > Any ideas? > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/30b41a83/attachment.html From 12ukwn at gmail.com Wed Jun 2 12:51:52 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 2 Jun 2010 21:51:52 +0200 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: <20100602192301.6e5710b3@anubis.defcon1> <891BEFD20CB841678F66A9EB11508C44@MOBILEE1705> Message-ID: <20100602215152.08e4c2fa@anubis.defcon1> Le Wed, 2 Jun 2010 15:36:19 -0400, "Madovsky" a ?crit : > We talk about justice. YOU talked about "justice"; lets talk about us justice: if you have a complain about a license, "justice" will only heard you IF you put $5M on the table - and only if you do so. > usually the victim complaints and the criminal must pay... > the contrary is called "mafia" -- I invoke Espy's law, which states that you all suck :P From delorenzodesign at gmail.com Wed Jun 2 12:53:17 2010 From: delorenzodesign at gmail.com (Michael De Lorenzo) Date: Wed, 2 Jun 2010 15:53:17 -0400 Subject: [Freeswitch-users] Set Privacy:id on Outbound Call Message-ID: I'm trying to set the PRIVACY SIP header to "id", but haven't been able to get it to set properly. I'm using this Lua statement: freeswitch.Session("{sip_cid_type=pid,origination_caller_id_name=Anonymous,origination_caller_id_number=+19725551234,privacy=full,ignore_early_media=true}sofia/gateway/mygw/19735555678}"); I've tried, sip_h_Privacy, too, but to no avail. On Wed, Jun 2, 2010 at 3:46 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. Status of absolute_codec_string (Kristian Kielhofner) > 2. Re: Using XLite/Softphone with Freeswitch2 (Michael De Lorenzo) > > > ---------- Forwarded message ---------- > From: Kristian Kielhofner > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 2 Jun 2010 15:40:44 -0400 > Subject: [Freeswitch-users] Status of absolute_codec_string > Hello everyone, > > Either I am misunderstanding absolute_codec_string or there is > something going on with it in trunk. I had a git checkout from a > month ago that exhibited this same behavior so I updated about an hour > ago. It's still there (or I'm still confused). > > In short, I have a call that comes in from a remote endpoint that > will advertise PCMU and G729 (in that order). When I bridge the call > I want to remove G729 capability from the outbound leg in some cases > so FS doesn't have to transcode. In other cases (controlled from the > dialplan), I want to place G729 as the first codec (again so FS > doesn't have to transcode) but I can force G729. I've tried regex on > ep_codec_string and I've tried regex on the SDP. Both seemed to work > well but what didn't work was setting absolute_codec_string from > either of these. > > Even when I try to set it manually without any fancy conditions it > doesn't appear to work. The codec prefs from the SIP profile are > presented to the remote endpoint. Full debug here: > > http://pastebin.freeswitch.org/13088 > > Any ideas? > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > > > > ---------- Forwarded message ---------- > From: Michael De Lorenzo > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 2 Jun 2010 15:45:38 -0400 > Subject: Re: [Freeswitch-users] Using XLite/Softphone with Freeswitch2 > Ok, what do I need to add to the dialplan? I haven't changed it at all > from the default. > > > On Wed, Jun 2, 2010 at 3:36 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: Using XLite/Softphone with Freeswitch (Brian West) >> 2. Re: Help get freeswitch.com back (Madovsky) >> 3. Re: Help get freeswitch.com back (Madovsky) >> 4. Re: Help get freeswitch.com back (Brian West) >> 5. Re: Help get freeswitch.com back (Anthony Minessale) >> 6. Re: Help get freeswitch.com back (Jean-Yves F. Barbier) >> 7. Re: Help get freeswitch.com back (Madovsky) >> >> >> ---------- Forwarded message ---------- >> From: Brian West >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 2 Jun 2010 14:02:30 -0500 >> Subject: Re: [Freeswitch-users] Using XLite/Softphone with Freeswitch >> 2010-06-02 14:49:56.853307 [INFO] mod_dialplan_xml.c:418 Processing John >> Doe->9735551234 in context public >> 2010-06-02 14:49:56.855781 [NOTICE] switch_core_state_machine.c:185 >> sofia/internal/9727289377 at 0.0.0.0 has executed the last dialplan >> instruction, hanging up. >> >> Its not matching anything in your dialplan. >> >> Press F8 >> >> /b >> >> On Jun 2, 2010, at 1:55 PM, Michael De Lorenzo wrote: >> >> > I'm trying to configure XLite as a softphone to use with our Freeswitch >> installation. It seems I can register successfully with FS, but if I try to >> dial someone (not another Freeswitch user, I haven't tried another >> Freeswitch user because I don't have any others and don't need that >> functionality) I receive a busy signal and a 480 Temporarily Unavailable >> response. >> > >> > From what I can tell in my SIP captures, the domain for the callee isn't >> getting properly set. I'm not sure where to set the domain (IP of my >> gateway) so the call can be completed. Note 0.0.0.0 is the IP address (I >> removed the actual IP) of my Freeswitch gateway and 1.1.1.1 is the IP >> address of the softphone/client. >> > >> > What setting am I missing? The "To" variable should be >> phonenumber at ip-of-my-gateway, the gateway I'm trying to use is the same >> one that works fine for outbound calls executed from the Freeswitch console. >> > >> > Can someone point me in the right direction? >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: "Madovsky" >> To: >> Date: Wed, 2 Jun 2010 15:04:47 -0400 >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> So why not send some beautiful network packets to these guys ? >> >> ----- Original Message ----- From: "Jean-Yves F. Barbier" < >> 12ukwn at gmail.com> >> To: >> Sent: Wednesday, June 02, 2010 2:08 PM >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> >> >> Le Wed, 2 Jun 2010 12:27:42 -0500, >>> Brian West a ?crit : >>> >>> To do that you have to pay to file a complaint and have someone handle >>>> the dispute for you. Its not cheap to do this the right way. >>>> >>> >>> what a wonderful us world...... >>> >>> -- >>> Good news. Ten weeks from Friday will be a pretty good day. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> >> >> ---------- Forwarded message ---------- >> From: "Madovsky" >> To: >> Date: Wed, 2 Jun 2010 15:08:06 -0400 >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> ICANN asks real admin info when you register a domain. >> in case of fake personal info the domain name should be removed. >> >> >> >> ----- Original Message ----- From: "Brian West" >> To: >> Sent: Wednesday, June 02, 2010 1:27 PM >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> >> >> To do that you have to pay to file a complaint and have someone handle >>> the dispute for you. Its not cheap to do this the right way. >>> >>> /b >>> >>> On Jun 2, 2010, at 12:23 PM, Jean-Yves F. Barbier wrote: >>> >>> so complain to the authority: they don't like very much fakd IDs. >>>> >>>> -- >>>> I wouldn't marry her with a ten foot pole. >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Brian West >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 2 Jun 2010 14:12:25 -0500 >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> Still takes money to file all the paperwork. Are you willing to do the >> stack of paperwork for that? >> >> >> /b >> >> On Jun 2, 2010, at 2:08 PM, Madovsky wrote: >> >> ICANN asks real admin info when you register a domain. >> in case of fake personal info the domain name should be removed. >> >> >> >> ----- Original Message ----- >> From: "Brian West" >> To: >> Sent: Wednesday, June 02, 2010 1:27 PM >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> >> >> To do that you have to pay to file a complaint and have someone handle the >> >> >> dispute for you. Its not cheap to do this the right way. >> >> >> /b >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 2 Jun 2010 14:19:59 -0500 >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> The only way to get it back for sure is to file the dispute with ICANN. >> Like I said in the original email, they charge 1300 to have one person >> decide on it and 2600 to have 3 people. >> Maybe they will sell it to us for less knowing that they will lose once we >> file the dispute but I doubt it. >> >> >> On Wed, Jun 2, 2010 at 2:04 PM, Madovsky wrote: >> >>> So why not send some beautiful network packets to these guys ? >>> >>> ----- Original Message ----- >>> From: "Jean-Yves F. Barbier" <12ukwn at gmail.com> >>> To: >>> Sent: Wednesday, June 02, 2010 2:08 PM >>> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >>> >>> >>> > Le Wed, 2 Jun 2010 12:27:42 -0500, >>> > Brian West a ?crit : >>> > >>> >> To do that you have to pay to file a complaint and have someone handle >>> >> the dispute for you. Its not cheap to do this the right way. >>> > >>> > what a wonderful us world...... >>> > >>> > -- >>> > Good news. Ten weeks from Friday will be a pretty good day. >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.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 >>> >> >> >> >> -- >> 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 >> >> >> ---------- Forwarded message ---------- >> From: "Jean-Yves F. Barbier" <12ukwn at gmail.com> >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 2 Jun 2010 21:23:05 +0200 >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> Le Wed, 2 Jun 2010 15:08:06 -0400, >> "Madovsky" a ?crit : >> >> > ICANN asks real admin info when you register a domain. >> > in case of fake personal info the domain name should be removed. >> >> Yeah, that's what I meant; especially if addresses are faked. >> >> -- >> Life is the childhood of our immortality. >> -- Goethe >> >> >> >> >> ---------- Forwarded message ---------- >> From: "Madovsky" >> To: >> Date: Wed, 2 Jun 2010 15:36:19 -0400 >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> We talk about justice. >> usually the victim complaints and the criminal must pay... >> the contrary is called "mafia" >> >> >> >> ----- Original Message ----- >> *From:* Brian West >> *To:* freeswitch-users at lists.freeswitch.org >> *Sent:* Wednesday, June 02, 2010 3:12 PM >> *Subject:* Re: [Freeswitch-users] Help get freeswitch.com back >> >> Still takes money to file all the paperwork. Are you willing to do the >> stack of paperwork for that? >> >> >> /b >> >> On Jun 2, 2010, at 2:08 PM, Madovsky wrote: >> >> ICANN asks real admin info when you register a domain. >> in case of fake personal info the domain name should be removed. >> >> >> >> ----- Original Message ----- >> From: "Brian West" >> To: >> Sent: Wednesday, June 02, 2010 1:27 PM >> Subject: Re: [Freeswitch-users] Help get freeswitch.com back >> >> >> To do that you have to pay to file a complaint and have someone handle the >> >> >> dispute for you. Its not cheap to do this the right way. >> >> >> /b >> >> >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/1d280d8c/attachment-0001.html From brian at freeswitch.org Wed Jun 2 12:55:00 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 14:55:00 -0500 Subject: [Freeswitch-users] Using XLite/Softphone with Freeswitch2 In-Reply-To: References: Message-ID: <2742C6F4-141B-494B-9E76-33E84B2E6B49@freeswitch.org> First off I have to say you need to not use Digest list subscription if you're going to be replying to messages as they'll get lost and not threaded properly. Secondly, yes you have to setup the dialplan to route the number you're trying to call to something... otherwise its going to hangup and not do anything because it has no route. /b On Jun 2, 2010, at 2:45 PM, Michael De Lorenzo wrote: > Ok, what do I need to add to the dialplan? I haven't changed it at all from the default. > > > On Wed, Jun 2, 2010 at 3:36 PM, 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: Using XLite/Softphone with Freeswitch (Brian West) > 2. Re: Help get freeswitch.com back (Madovsky) > 3. Re: Help get freeswitch.com back (Madovsky) > 4. Re: Help get freeswitch.com back (Brian West) > 5. Re: Help get freeswitch.com back (Anthony Minessale) > 6. Re: Help get freeswitch.com back (Jean-Yves F. Barbier) > 7. Re: Help get freeswitch.com back (Madovsky) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/b6670e30/attachment.html From kris at kriskinc.com Wed Jun 2 13:06:29 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 2 Jun 2010 16:06:29 -0400 Subject: [Freeswitch-users] Status of absolute_codec_string In-Reply-To: <584D7DCB-37BA-4E95-9CEF-719012B6CEFF@freeswitch.org> References: <584D7DCB-37BA-4E95-9CEF-719012B6CEFF@freeswitch.org> Message-ID: Brian, late-negotiation is set. I tried inherit_codec and while it does what you say it can't handle cases where the endpoints are incompatible: A leg (PCMU only) -> B (PCMA only) = INCOMPATIBLE_DESTINATION My method would allow me to *always* present at least one compatible codec AND place it at the bottom of the SDP offer. If the B leg happens to be compatible with more preferred codecs it can use them with no transcoding. This is all (of course) assuming I can set this using absolute_codec_string and FS will transcode for me across these two channels using my common codec when required... On Wed, Jun 2, 2010 at 3:48 PM, Brian West wrote: > You can enable "late-negotiation", then set inherit_codec=true then it will > delay the negotiation till it gets the answer from the B-Leg before the > A-Leg's codec is thus accomplishing the same thing. > /b > > On Jun 2, 2010, at 2:40 PM, Kristian Kielhofner wrote: > > Hello everyone, > > ?Either I am misunderstanding absolute_codec_string or there is > something going on with it in trunk. ?I had a git checkout from a > month ago that exhibited this same behavior so I updated about an hour > ago. ?It's still there (or I'm still confused). > > ?In short, I have a call that comes in from a remote endpoint that > will advertise PCMU and G729 (in that order). ?When I bridge the call > I want to remove G729 capability from the outbound leg in some cases > so FS doesn't have to transcode. ?In other cases (controlled from the > dialplan), I want to place G729 as the first codec (again so FS > doesn't have to transcode) but I can force G729. ?I've tried regex on > ep_codec_string and I've tried regex on the SDP. ?Both seemed to work > well but what didn't work was setting absolute_codec_string from > either of these. > > ?Even when I try to set it manually without any fancy conditions it > doesn't appear to work. ?The codec prefs from the SIP profile are > presented to the remote endpoint. ?Full debug here: > > http://pastebin.freeswitch.org/13088 > > ?Any ideas? > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From infos at madovsky.org Wed Jun 2 13:08:50 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 2 Jun 2010 16:08:50 -0400 Subject: [Freeswitch-users] Help get freeswitch.com back References: <20100602192301.6e5710b3@anubis.defcon1><891BEFD20CB841678F66A9EB11508C44@MOBILEE1705> <20100602215152.08e4c2fa@anubis.defcon1> Message-ID: I used the word justice philosophically. of course the real world we live is not justice... ----- Original Message ----- From: "Jean-Yves F. Barbier" <12ukwn at gmail.com> To: Sent: Wednesday, June 02, 2010 3:51 PM Subject: Re: [Freeswitch-users] Help get freeswitch.com back > Le Wed, 2 Jun 2010 15:36:19 -0400, > "Madovsky" a ?crit : > > >> We talk about justice. > > YOU talked about "justice"; lets talk about us justice: if you have a > complain about a license, "justice" will only heard you IF you put $5M on > the table - and only if you do so. > >> usually the victim complaints and the criminal must pay... >> the contrary is called "mafia" > > -- > I invoke Espy's law, which states that you all suck :P > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 david.ponzone at gmail.com Wed Jun 2 13:23:59 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Wed, 2 Jun 2010 22:23:59 +0200 Subject: [Freeswitch-users] Set Privacy:id on Outbound Call In-Reply-To: References: Message-ID: Michael, Can you please send a new message and not reply to the digest ? To set Privacy:id, use: sip_h_Privacy=id Regardsn David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/43c886ff/attachment.html From ranjtech at gmail.com Wed Jun 2 14:17:31 2010 From: ranjtech at gmail.com (RR) Date: Wed, 2 Jun 2010 17:17:31 -0400 Subject: [Freeswitch-users] Direct inward dialling In-Reply-To: References: <491B305B.4060307@gmx.net> <072A175C-9004-4C14-90EC-9A93A8453787@gmail.com> <35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org> Message-ID: On Wed, Jun 2, 2010 at 2:17 PM, Michael Collins wrote: > > > On Tue, Jun 1, 2010 at 7:57 PM, Ron McLeod wrote: > >> Is [32|48|54|55|65] correct? Shouldn?t it be (32|48|54|55|65) >> instead? >> >> >> >> >> >> ^\+?1?(0[0-1]+)?((32|48|54|55|65)\d+)\;?(phone-context=)?\+?(\d+)?$ >> > Also, you really don't need the dash in [0-1], just do [01] which means > "match a 0 or a 1" > -MC > > Ok cool. Thanks for that. BTW, I've noticed that the regex CLI command doesn't always behave the same way as the core FS engine. A lot of tests that display a different result when testing using the regex command yield different behaviour when one makes a call and how FS behaves/treats the call. Maybe it's a bug in the old version. We're using a very old version of FS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/7cfe0e78/attachment-0001.html From pray at theprossergroup.com Wed Jun 2 14:29:36 2010 From: pray at theprossergroup.com (Praveen Ray) Date: Wed, 2 Jun 2010 17:29:36 -0400 Subject: [Freeswitch-users] No Early Media Ringback In-Reply-To: <4C0675F5.2050907@aktzero.com> References: <4C0675F5.2050907@aktzero.com> Message-ID: Thanks Andrew Could you please point me to the right place (xml file) where should be placed? -P On Wed, Jun 2, 2010 at 11:17 AM, Andrew Thompson wrote: > On 6/2/2010 8:11 AM, Praveen Ray wrote: > > Hi All > > I am using Freeswitch 1.0.6 with Vitelity as inbound/outbound > > provider. My Inbound calls route properly to extension 1000, however, > > the caller does not hear the extension ringing. The extension at 1000 > > does ring however and once I pick up the call, the conversation > > proceeds normally. I just can't figure out why the caller hears > > silence during phone ringing. I have tried putting instant_ringback in > > dialplan/public.xml: > > I get this from a couple VOIP providers. My main number goes straight to > a IVR so it's a little strange for some callers, but for the others > ring_ready seems to help. If you immediately start doing stuff, like an > IVR, it won't help unless you put in a delay(so the ringing can happen) > before you answer. But the way I see it, once you've got the call, do > something with it! If the inbound call just rings until someone answers, > this should do it: > > > > > -- > Andrew Thompson > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/37825ffc/attachment.html From infos at madovsky.org Wed Jun 2 14:38:57 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 2 Jun 2010 17:38:57 -0400 Subject: [Freeswitch-users] No Early Media Ringback References: <4C0675F5.2050907@aktzero.com> Message-ID: <8F46981695334320AAD64327AC60183A@MOBILEE1705> in a dialplan ----- Original Message ----- From: Praveen Ray To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, June 02, 2010 5:29 PM Subject: Re: [Freeswitch-users] No Early Media Ringback Thanks Andrew Could you please point me to the right place (xml file) where should be placed? -P On Wed, Jun 2, 2010 at 11:17 AM, Andrew Thompson wrote: On 6/2/2010 8:11 AM, Praveen Ray wrote: > Hi All > I am using Freeswitch 1.0.6 with Vitelity as inbound/outbound > provider. My Inbound calls route properly to extension 1000, however, > the caller does not hear the extension ringing. The extension at 1000 > does ring however and once I pick up the call, the conversation > proceeds normally. I just can't figure out why the caller hears > silence during phone ringing. I have tried putting instant_ringback in > dialplan/public.xml: I get this from a couple VOIP providers. My main number goes straight to a IVR so it's a little strange for some callers, but for the others ring_ready seems to help. If you immediately start doing stuff, like an IVR, it won't help unless you put in a delay(so the ringing can happen) before you answer. But the way I see it, once you've got the call, do something with it! If the inbound call just rings until someone answers, this should do it: -- Andrew Thompson _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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/20100602/215815ac/attachment.html From gavin.henry at gmail.com Wed Jun 2 15:01:53 2010 From: gavin.henry at gmail.com (Gavin Henry) Date: Wed, 2 Jun 2010 23:01:53 +0100 Subject: [Freeswitch-users] OSTAG - Open Source Telephony Action Group In-Reply-To: References: Message-ID: On 1 June 2010 20:20, Michael Collins wrote: > Good news! > > The Open Source Telephony Action Group (OSTAG) is almost ready to become an > official 501(c)3 non-profit organization. We need to raise $800 in order to > file the necessary paperwork. We have set up a PayPal account where we can > all send our donations: > > donations at ostag.org > > Please assist us in moving this important organization forward. OSTAG is > dedicated to the advancement of open source telecommuncations software that > improves the lives of people all over the world. This is a most worthy > cause! Please help now and in the future. > I presume this is US based? -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From brian at freeswitch.org Wed Jun 2 15:06:27 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jun 2010 17:06:27 -0500 Subject: [Freeswitch-users] OSTAG - Open Source Telephony Action Group In-Reply-To: References: Message-ID: <41BEAC63-2E0E-4E31-84C6-1CCD14D3DF8E@freeswitch.org> Yes it is. Once it gets established then I suspect the organization could apply for non-profit in other countries. /b On Jun 2, 2010, at 5:01 PM, Gavin Henry wrote: > On 1 June 2010 20:20, Michael Collins wrote: >> Good news! >> >> The Open Source Telephony Action Group (OSTAG) is almost ready to become an >> official 501(c)3 non-profit organization. We need to raise $800 in order to >> file the necessary paperwork. We have set up a PayPal account where we can >> all send our donations: >> >> donations at ostag.org >> >> Please assist us in moving this important organization forward. OSTAG is >> dedicated to the advancement of open source telecommuncations software that >> improves the lives of people all over the world. This is a most worthy >> cause! Please help now and in the future. >> > > I presume this is US based? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/fb9a3249/attachment.html From delorenzodesign at gmail.com Wed Jun 2 16:00:20 2010 From: delorenzodesign at gmail.com (Michael De Lorenzo) Date: Wed, 2 Jun 2010 19:00:20 -0400 Subject: [Freeswitch-users] Set Privacy:id on Outbound Call Message-ID: Does this conflict with setting the originator caller id and name? Sorry for replying to the full digest. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/b8333ec0/attachment.html From delorenzodesign at gmail.com Wed Jun 2 16:02:30 2010 From: delorenzodesign at gmail.com (Michael De Lorenzo) Date: Wed, 2 Jun 2010 19:02:30 -0400 Subject: [Freeswitch-users] Using XLite/Softphone with Freeswitch Message-ID: Is there a best practice for organizing these dialplans? Or is it more the developer's preference? Apologies for replying to the full digest, it was a mistake. .michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100602/9282933c/attachment.html From david.ponzone at gmail.com Wed Jun 2 16:18:07 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 3 Jun 2010 01:18:07 +0200 Subject: [Freeswitch-users] Set Privacy:id on Outbound Call In-Reply-To: References: Message-ID: <7109E260-67DB-4DEA-BFF8-7912C4193BB2@gmail.com> Conflict ? Not really, it's 2 different things. You can send an INVITE with a valid CLID and with Privacy:id. Everything will depend on your carrier. I guess your carrier supports Privacy field. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 03/06/2010 ? 01:00, Michael De Lorenzo a ?crit : > Does this conflict with setting the originator caller id and name? > > Sorry for replying to the full digest. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/98a04716/attachment-0001.html From bwibowo at gmail.com Wed Jun 2 16:26:21 2010 From: bwibowo at gmail.com (Budi wibowo) Date: Wed, 2 Jun 2010 23:26:21 +0000 Subject: [Freeswitch-users] Freeswicth nms Message-ID: <895033401-1275521270-cardhu_decombobulator_blackberry.rim.net-328548762-@bda057.bisx.prodap.on.blackberry> Hi all I'm looking for nms that can do minimum alarm,fault, performance and statistic. Alarm could be sent via sms or email. I need the such solution due I have project to involve more than 10 fs box. Both free and commercial solution are ok Regards Budi From anthony.minessale at gmail.com Wed Jun 2 18:23:13 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Jun 2010 20:23:13 -0500 Subject: [Freeswitch-users] Status of absolute_codec_string In-Reply-To: References: <584D7DCB-37BA-4E95-9CEF-719012B6CEFF@freeswitch.org> Message-ID: you have to export it instead of set to present it both directions otherwise you need to put it in the {} to present it On Wed, Jun 2, 2010 at 3:06 PM, Kristian Kielhofner wrote: > Brian, > > late-negotiation is set. > > I tried inherit_codec and while it does what you say it can't handle > cases where the endpoints are incompatible: > > A leg (PCMU only) -> B (PCMA only) = INCOMPATIBLE_DESTINATION > > My method would allow me to *always* present at least one compatible > codec AND place it at the bottom of the SDP offer. If the B leg > happens to be compatible with more preferred codecs it can use them > with no transcoding. This is all (of course) assuming I can set this > using absolute_codec_string and FS will transcode for me across these > two channels using my common codec when required... > > On Wed, Jun 2, 2010 at 3:48 PM, Brian West wrote: > > You can enable "late-negotiation", then set inherit_codec=true then it > will > > delay the negotiation till it gets the answer from the B-Leg before the > > A-Leg's codec is thus accomplishing the same thing. > > /b > > > > On Jun 2, 2010, at 2:40 PM, Kristian Kielhofner wrote: > > > > Hello everyone, > > > > Either I am misunderstanding absolute_codec_string or there is > > something going on with it in trunk. I had a git checkout from a > > month ago that exhibited this same behavior so I updated about an hour > > ago. It's still there (or I'm still confused). > > > > In short, I have a call that comes in from a remote endpoint that > > will advertise PCMU and G729 (in that order). When I bridge the call > > I want to remove G729 capability from the outbound leg in some cases > > so FS doesn't have to transcode. In other cases (controlled from the > > dialplan), I want to place G729 as the first codec (again so FS > > doesn't have to transcode) but I can force G729. I've tried regex on > > ep_codec_string and I've tried regex on the SDP. Both seemed to work > > well but what didn't work was setting absolute_codec_string from > > either of these. > > > > Even when I try to set it manually without any fancy conditions it > > doesn't appear to work. The codec prefs from the SIP profile are > > presented to the remote endpoint. Full debug here: > > > > http://pastebin.freeswitch.org/13088 > > > > Any ideas? > > > > -- > > Kristian Kielhofner > > http://www.astlinux.org > > http://blog.krisk.org > > http://www.star2star.com > > http://www.submityoursip.com > > http://www.voalte.com > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100602/217d140c/attachment.html From mrene_lists at avgs.ca Wed Jun 2 22:54:24 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 3 Jun 2010 01:54:24 -0400 Subject: [Freeswitch-users] Status of absolute_codec_string In-Reply-To: References: <584D7DCB-37BA-4E95-9CEF-719012B6CEFF@freeswitch.org> Message-ID: <888BC5A7-5517-47E1-8519-4DC8E70C463C@avgs.ca> When you have late negotiation on, the ep_codec_string variable will contained the parsed SDP in the same format as needed by absolute_codec_string, you can therefore set absolute_codec_string=${ep_codec_string} and inherit_codec=true in order to offer all the A-leg codecs on the B-leg. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-06-02, at 9:23 PM, Anthony Minessale wrote: > you have to export it instead of set to present it both directions > otherwise you need to put it in the {} to present it > > On Wed, Jun 2, 2010 at 3:06 PM, Kristian Kielhofner wrote: > Brian, > > late-negotiation is set. > > I tried inherit_codec and while it does what you say it can't handle > cases where the endpoints are incompatible: > > A leg (PCMU only) -> B (PCMA only) = INCOMPATIBLE_DESTINATION > > My method would allow me to *always* present at least one compatible > codec AND place it at the bottom of the SDP offer. If the B leg > happens to be compatible with more preferred codecs it can use them > with no transcoding. This is all (of course) assuming I can set this > using absolute_codec_string and FS will transcode for me across these > two channels using my common codec when required... > > On Wed, Jun 2, 2010 at 3:48 PM, Brian West wrote: > > You can enable "late-negotiation", then set inherit_codec=true then it will > > delay the negotiation till it gets the answer from the B-Leg before the > > A-Leg's codec is thus accomplishing the same thing. > > /b > > > > On Jun 2, 2010, at 2:40 PM, Kristian Kielhofner wrote: > > > > Hello everyone, > > > > Either I am misunderstanding absolute_codec_string or there is > > something going on with it in trunk. I had a git checkout from a > > month ago that exhibited this same behavior so I updated about an hour > > ago. It's still there (or I'm still confused). > > > > In short, I have a call that comes in from a remote endpoint that > > will advertise PCMU and G729 (in that order). When I bridge the call > > I want to remove G729 capability from the outbound leg in some cases > > so FS doesn't have to transcode. In other cases (controlled from the > > dialplan), I want to place G729 as the first codec (again so FS > > doesn't have to transcode) but I can force G729. I've tried regex on > > ep_codec_string and I've tried regex on the SDP. Both seemed to work > > well but what didn't work was setting absolute_codec_string from > > either of these. > > > > Even when I try to set it manually without any fancy conditions it > > doesn't appear to work. The codec prefs from the SIP profile are > > presented to the remote endpoint. Full debug here: > > > > http://pastebin.freeswitch.org/13088 > > > > Any ideas? > > > > -- > > Kristian Kielhofner > > http://www.astlinux.org > > http://blog.krisk.org > > http://www.star2star.com > > http://www.submityoursip.com > > http://www.voalte.com > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/4b5b27fe/attachment.html From david.ponzone at gmail.com Wed Jun 2 23:23:23 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 3 Jun 2010 08:23:23 +0200 Subject: [Freeswitch-users] Status of absolute_codec_string In-Reply-To: References: <584D7DCB-37BA-4E95-9CEF-719012B6CEFF@freeswitch.org> Message-ID: Kristian, I am not sure to understand how removing G729 from PCMU/G729 would allow a call to a PCMA device... That was the example you gave in your first mail. I think you would need to at least present PCMU/PCMA on all calls. Anyway, setting absolute_codec_string works perfectly here. But as Anthony already told you, you got to export it (it affects B- leg). David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 02/06/2010 ? 22:06, Kristian Kielhofner a ?crit : > Brian, > > late-negotiation is set. > > I tried inherit_codec and while it does what you say it can't handle > cases where the endpoints are incompatible: > > A leg (PCMU only) -> B (PCMA only) = INCOMPATIBLE_DESTINATION > > My method would allow me to *always* present at least one compatible > codec AND place it at the bottom of the SDP offer. If the B leg > happens to be compatible with more preferred codecs it can use them > with no transcoding. This is all (of course) assuming I can set this > using absolute_codec_string and FS will transcode for me across these > two channels using my common codec when required... > > On Wed, Jun 2, 2010 at 3:48 PM, Brian West > wrote: >> You can enable "late-negotiation", then set inherit_codec=true then >> it will >> delay the negotiation till it gets the answer from the B-Leg before >> the >> A-Leg's codec is thus accomplishing the same thing. >> /b >> >> On Jun 2, 2010, at 2:40 PM, Kristian Kielhofner wrote: >> >> Hello everyone, >> >> Either I am misunderstanding absolute_codec_string or there is >> something going on with it in trunk. I had a git checkout from a >> month ago that exhibited this same behavior so I updated about an >> hour >> ago. It's still there (or I'm still confused). >> >> In short, I have a call that comes in from a remote endpoint that >> will advertise PCMU and G729 (in that order). When I bridge the call >> I want to remove G729 capability from the outbound leg in some cases >> so FS doesn't have to transcode. In other cases (controlled from the >> dialplan), I want to place G729 as the first codec (again so FS >> doesn't have to transcode) but I can force G729. I've tried regex on >> ep_codec_string and I've tried regex on the SDP. Both seemed to work >> well but what didn't work was setting absolute_codec_string from >> either of these. >> >> Even when I try to set it manually without any fancy conditions it >> doesn't appear to work. The codec prefs from the SIP profile are >> presented to the remote endpoint. Full debug here: >> >> http://pastebin.freeswitch.org/13088 >> >> Any ideas? >> >> -- >> Kristian Kielhofner >> http://www.astlinux.org >> http://blog.krisk.org >> http://www.star2star.com >> http://www.submityoursip.com >> http://www.voalte.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/9b3e45fb/attachment-0001.html From shaheryarkh at googlemail.com Wed Jun 2 23:23:44 2010 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Thu, 3 Jun 2010 11:23:44 +0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: Message-ID: Unfortunately Paypal is not supported in Pakistan. Do you have any other supported medium, e.g. Western Union or direct credit card payment? etc. Thank you. On Wed, Jun 2, 2010 at 8:57 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Some squatter has had freeswitch.com for years and it's time we had it > back. > > We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party > panel to resolve the dispute. > They are using the domain to pose as a VoIP site (if you keep loading > http://www.freeswitch.com/ you can see it in action) > > Don't let creeps like this misuse the internet! Use the paypal button on > the site to donate to the cause. > > Everyone who donates will get their name up on our thank you page on our > site. > Put "freeswitch.com UDRP and your name as you would like to see it on the > thank-you page" in the note on the paypal form. > > > > -- > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 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/20100603/46d9c4f8/attachment.html From gustavo.espeche at upper-soft.com Thu Jun 3 02:39:53 2010 From: gustavo.espeche at upper-soft.com (Gustavo Espeche) Date: Thu, 03 Jun 2010 06:39:53 -0300 Subject: [Freeswitch-users] xml_curl endpoint register In-Reply-To: References: <20100602105026.fkxak9i0pw0c0c4k@www.upper-soft.com> Message-ID: <1275557993.2200.21.camel@gustavo-laptop> Hi Michael thank for your quick answer, well i develop a littel php script that register and endpoint with FS through xml_curl module, but some endpoint like xlite or pangolin(open port 8731 ) don't open the port 5060 for receive sip calls. Follow the row post that i receive when and enpoint try to register with fs Post hostname=server2§ion=directory&tag_name=domain&key_name=name&key_value=10.0.0.101&Event-Name=REQUEST_PARAMS&Core-UUID=41841292-6d8d-11df-b413-6bc960363dcb&Free SWITCH-Hostname=server2&FreeSWITCH-IPv4=10.0.0.101&FreeSWITCH-IPv6=%3A% 3A1&Event-Date-Local=2010-06-01%2011%3A56%3A55&Event-Date-GMT=Tue,%2001% 20Jun%202010%2014%3A56%3A 55% 20GMT&Event-Date-Timestamp=1275404215713110&Event-Calling-File=sofia_reg.c&Event-Calling-Function=sofia_reg_parse_auth&Event-Calling-Line-Number=1804&action=sip_auth &sip_profile=internal&sip_user_agent=PortGo%20v6.0,%20Build% 2005202010&sip_auth_username=gustavo2&sip_auth_realm=10.0.0.101&sip_auth_nonce=eba2ee24-6d8d-11df-b425-6bc96 0363dcb&sip_auth_uri=sip% 3A10.0.0.101&sip_contact_user=gustavo2&sip_contact_host=10.0.0.15&sip_to_user=gustavo2&sip_to_host=10.0.0.101&sip_from_user=gustavo2&sip_from_h ost=10.0.0.101&sip_request_host=10.0.0.101&sip_auth_qop=auth&sip_auth_cnonce=454c337c7d03b616fb964334cc72e260&sip_auth_nc=00000001&sip_auth_response=a436fcd15330a644adc 7d07f44684b55&sip_auth_method=REGISTER&key=id&user=gustavo2&domain=10.0.0.101&ip=10.0.0.15 in this post i can't find the endpoint sip port, and i can't register in my db what is the sip port of this endpoint.I have a php routing script too, and when some endpoint try to call to one endpoint that don't use 5060 as sip port, i haven't the information in my database to give the right sofia bridge route. this is my routing script answer when someone dial gustavo2
the right answer is the ip of the endpoint:port but in registration i don't have the local sip port of the endpoint
maybe i'm using a wrong way to dial between endpoint in FS, i using this way because our system is oriented to wholesale market and the most of route are external gw, and we try to have the same routing script for register endpoint and gw not register Best Regard On Wed, 2010-06-02 at 11:40 -0700, Michael Collins wrote: > Welcome to FreeSWITCH!8057 > > I think you'll find a lot of helpful folks around here. Also, don't > forget to join the IRC channel: #freeswitch on irc.freenode.net where > you can discuss things in realtime. > > I'm having trouble visualizing what you are doing. Can you expand your > description of what you are doing? You might want to put some logs > into pastebin.freeswitch.org and link to those logs in this thread. > > Thanks, > MC > > On Wed, Jun 2, 2010 at 6:50 AM, Gustavo Espeche > wrote: > Hello all, i'm new in freeswitch we are migrating our software > form > yate to freeswitch. > I'm writing a registration php script using xml_curl module, > the > registration was good but some endpoint don't use 5060 as sip > port and > when i register it, i can't find the endpoint port in curl > post, > because of then when i dial to an endpoint that isn't listen > in sip > port 5060 the call fail. > Some one know how can i fix it? > when i register the endpoint i insert in a mysql db time, ip, > and when > some one dial to this user i read my db for know the user's > ip. > but i need insert the endpoint sip's port too for it work. > > Best Regards. > > Gustavo Espeche > www.easyipcall.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 codecomplete at free.fr Thu Jun 3 05:48:18 2010 From: codecomplete at free.fr (GillesToo) Date: Thu, 3 Jun 2010 05:48:18 -0700 (PDT) Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: Message-ID: <1275569298280-5134985.post@n2.nabble.com> Anthony Minessale wrote: > Some squatter has had freeswitch.com for years and it's time we had it > back. I suggess updating the homepage so that everyone who happens on the site knows that they could contribute to getting the .COM back -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-get-freeswitch-com-back-tp5131459p5134985.html Sent from the freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Jun 3 05:52:00 2010 From: codecomplete at free.fr (GillesToo) Date: Thu, 3 Jun 2010 05:52:00 -0700 (PDT) Subject: [Freeswitch-users] freeSWITCH segfault on openwrt In-Reply-To: References: Message-ID: <1275569520907-5135008.post@n2.nabble.com> Woody Dickson wrote: > I manage to get freeSWITCH to compile and run on openwrt, but then > freeSWITCH keeps crashing when the number of calls reaches 25 > consistently. Out of curiosity, on what hardware do you run OpenWrt + Freeswitch? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeSWITCH-segfault-on-openwrt-tp5008807p5135008.html Sent from the freeswitch-users mailing list archive at Nabble.com. From michaelt at voxcore.voxtelecom.co.za Thu Jun 3 06:09:19 2010 From: michaelt at voxcore.voxtelecom.co.za (Michael Toop) Date: Thu, 3 Jun 2010 15:09:19 +0200 Subject: [Freeswitch-users] xml_cdr http post differentiation between legs In-Reply-To: <034401caba3e$70bbb950$52332bf0$@net> References: <034401caba3e$70bbb950$52332bf0$@net> Message-ID: Hi Adam, We also have this problem (no way to tell the a-leg from the b-leg in xml_cdr POST data). What I do, which is a bit of a work around, is to look for the typical SIP media IP of the b-leg of the call to differentiate the a from b legs. It would however be great to have a tag in the POST to make life easier, similar to the feature where xml_cdr file is prefixed with a 'a_'. Cheers, Michael On Tue, Mar 2, 2010 at 9:27 PM, Adam Ford wrote: > Is there any way to tell whether it is the a-leg or b-leg when receiving > the POST data from mod_xml_cdr? > > > > I wrote a parser in Python which parses the a_*.cdr.xml log files, and only > grabs the info in the associated b-leg log files if it needs additional > information(for example - in the case of a call that ends with mod_fifo). I > want to adapt this to using the POST features of mod_xml_cdr so that it can > be real time. > > > > Any suggestions are greatly appreciated, > > -Adam > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/9e5ba76d/attachment.html From suvirkumar at gmail.com Wed Jun 2 18:00:38 2010 From: suvirkumar at gmail.com (suvirkumar at gmail.com) Date: Thu, 3 Jun 2010 01:00:38 +0000 Subject: [Freeswitch-users] Freeswicth nms Message-ID: <1111639352-1275526834-cardhu_decombobulator_blackberry.rim.net-2040530345-@bda2058.bisx.prodap.on.blackberry> Budi, Try Nagios for network monitoring Regards Suvir ------Original Message------ From: Budi wibowo Sender: freeswitch-users-bounces at lists.freeswitch.org To: freeswitch-users at lists.freeswitch.org ReplyTo: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswicth nms Sent: Jun 3, 2010 06:26 Hi all I'm looking for nms that can do minimum alarm,fault, performance and statistic. Alarm could be sent via sms or email. I need the such solution due I have project to involve more than 10 fs box. Both free and commercial solution are ok Regards Budi _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Sent via BlackBerry? from AIS From linuxwhy at 163.com Thu Jun 3 01:49:36 2010 From: linuxwhy at 163.com (=?GB2312?B?zfW66dPA?=) Date: Thu, 3 Jun 2010 16:49:36 +0800 Subject: [Freeswitch-users] I edit a china tones.conf , but it's not useful. does anybody can help me? Message-ID: <20100603164936.49806012@honiswang-laptop> I edit a china tones.conf ,but it's not useful. does anybody can help me? [cn] generate-dial => v=-7;%(1000,0,475) detect-dial => 475 generate-ring => v=-7;%(1000,4000,475) detect-ring => 475 generate-busy => v=-7;%(350,350,475) detect-busy => 475 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440,480 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 1776.7 tones for china China (People?s Republic of) Busy tone - 450 0.35 on 0.35 off Congestion tone - 450 0.7 on 0.7 off Dial tone - 450 continuous Second dial tone - 450 continuous Intrusion tone - 450 0.2 on 0.2 off 0.2 on 0.6 off Number unobtainable tone - 450 3x(0.1 on 0.1 off) 0.4 on 0.4 off Ringing tone - 450 1.0 on 4.0 off Special information tone - I 450 0.4 on 0.04 off Special information tone - II 950 0.4 on 10.0 off Waiting tone - 450 0.4 on 4.0 off ??? ????-IP??? Thu, 3 Jun 2010 16:45:17 +0800o From kris at kriskinc.com Thu Jun 3 08:04:18 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 3 Jun 2010 11:04:18 -0400 Subject: [Freeswitch-users] Status of absolute_codec_string In-Reply-To: References: <584D7DCB-37BA-4E95-9CEF-719012B6CEFF@freeswitch.org> Message-ID: That was it! I realized I'd only used absolute_codec_string with bridge prior to this. Thanks a lot! On Wed, Jun 2, 2010 at 9:23 PM, Anthony Minessale wrote: > you have to export it instead of set to present it both directions > otherwise you need to put it in the {} to present it -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From edpimentl at gmail.com Thu Jun 3 08:13:26 2010 From: edpimentl at gmail.com (EdPimentl) Date: Thu, 3 Jun 2010 11:13:26 -0400 Subject: [Freeswitch-users] Freeswicth nms In-Reply-To: <895033401-1275521270-cardhu_decombobulator_blackberry.rim.net-328548762-@bda057.bisx.prodap.on.blackberry> References: <895033401-1275521270-cardhu_decombobulator_blackberry.rim.net-328548762-@bda057.bisx.prodap.on.blackberry> Message-ID: Nagios + Splunk -E http://vCardCloud.com On Wed, Jun 2, 2010 at 7:26 PM, Budi wibowo wrote: > Hi all > I'm looking for nms that can do minimum alarm,fault, performance and > statistic. Alarm could be sent via sms or email. > I need the such solution due I have project to involve more than 10 fs box. > Both free and commercial solution are ok > > Regards > Budi > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/7e409351/attachment.html From anthony.minessale at gmail.com Thu Jun 3 08:29:52 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Jun 2010 10:29:52 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <1275569298280-5134985.post@n2.nabble.com> References: <1275569298280-5134985.post@n2.nabble.com> Message-ID: good idea, Michael??? On Thu, Jun 3, 2010 at 7:48 AM, GillesToo wrote: > > > Anthony Minessale wrote: > > Some squatter has had freeswitch.com for years and it's time we had it > > back. > > I suggess updating the homepage so that everyone who happens on the site > knows that they could contribute to getting the .COM back > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Help-get-freeswitch-com-back-tp5131459p5134985.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100603/5663e028/attachment.html From anatoliy at kounitskiy.com Thu Jun 3 08:33:59 2010 From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy) Date: Thu, 03 Jun 2010 18:33:59 +0300 Subject: [Freeswitch-users] Freeswicth nms In-Reply-To: <895033401-1275521270-cardhu_decombobulator_blackberry.rim.net-328548762-@bda057.bisx.prodap.on.blackberry> References: <895033401-1275521270-cardhu_decombobulator_blackberry.rim.net-328548762-@bda057.bisx.prodap.on.blackberry> Message-ID: <1275579239.1789.7.camel@lenovo400> At work, I use ZenOSS+custom bash scripts+monit for monitoring freeswitch servers. Regards, Anatoliy On Wed, 2010-06-02 at 23:26 +0000, Budi wibowo wrote: > Hi all > I'm looking for nms that can do minimum alarm,fault, performance and statistic. Alarm could be sent via sms or email. > I need the such solution due I have project to involve more than 10 fs box. > Both free and commercial solution are ok > > Regards > Budi > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy at kounitskiy.com Mobile: +359898913540 From steveayre at gmail.com Thu Jun 3 08:49:02 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 3 Jun 2010 16:49:02 +0100 Subject: [Freeswitch-users] xml_cdr http post differentiation between legs In-Reply-To: References: <034401caba3e$70bbb950$52332bf0$@net> Message-ID: Check the contents of the direction tag. It will be either inbound (aleg) or outbound (bleg). Steve on iPhone On 3 Jun 2010, at 14:09, Michael Toop wrote: > Hi Adam, > > We also have this problem (no way to tell the a-leg from the b-leg > in xml_cdr POST data). > > What I do, which is a bit of a work around, is to look for the > typical SIP media IP of the b-leg of the call to differentiate the a > from b legs. > > It would however be great to have a tag in the POST to make life > easier, similar to the feature where xml_cdr file is prefixed with a > 'a_'. > > Cheers, > > Michael > > > > On Tue, Mar 2, 2010 at 9:27 PM, Adam Ford wrote: > Is there any way to tell whether it is the a-leg or b-leg when > receiving the POST data from mod_xml_cdr? > > > > I wrote a parser in Python which parses the a_*.cdr.xml log files, > and only grabs the info in the associated b-leg log files if it > needs additional information(for example - in the case of a call > that ends with mod_fifo). I want to adapt this to using the POST > features of mod_xml_cdr so that it can be real time. > > > > Any suggestions are greatly appreciated, > > -Adam > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.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/20100603/b9907abb/attachment.html From msc at freeswitch.org Thu Jun 3 08:51:38 2010 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 3 Jun 2010 08:51:38 -0700 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: <1275569298280-5134985.post@n2.nabble.com> Message-ID: Definitely a good idea. When I get back in front of a computer I will work it up. Sent from my iPhone On Jun 3, 2010, at 8:29 AM, Anthony Minessale wrote: > good idea, Michael??? > > On Thu, Jun 3, 2010 at 7:48 AM, GillesToo > wrote: > > > Anthony Minessale wrote: > > Some squatter has had freeswitch.com for years and it's time we > had it > > back. > > I suggess updating the homepage so that everyone who happens on the > site > knows that they could contribute to getting the .COM back > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-get-freeswitch-com-back-tp5131459p5134985.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/61b8f3f0/attachment-0001.html From brian at freeswitch.org Thu Jun 3 08:56:42 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Jun 2010 10:56:42 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: <1275569298280-5134985.post@n2.nabble.com> Message-ID: <4535EF21-F837-456C-A167-82A8D914DC05@freeswitch.org> Its time for an iPad. /b On Jun 3, 2010, at 10:51 AM, Michael S Collins wrote: > Definitely a good idea. When I get back in front of a computer I will work it up. > > Sent from my iPhone From infos at madovsky.org Thu Jun 3 09:33:54 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 3 Jun 2010 12:33:54 -0400 Subject: [Freeswitch-users] Help get freeswitch.com back References: <1275569298280-5134985.post@n2.nabble.com> <4535EF21-F837-456C-A167-82A8D914DC05@freeswitch.org> Message-ID: <59C637855C5146D28EDE676D7A81405D@MOBILEE1705> didn't know FS team were a pro apple ;) hard to put an iPad in your pocket :D ----- Original Message ----- From: "Brian West" To: Sent: Thursday, June 03, 2010 11:56 AM Subject: Re: [Freeswitch-users] Help get freeswitch.com back > Its time for an iPad. > > /b > > On Jun 3, 2010, at 10:51 AM, Michael S Collins wrote: > >> Definitely a good idea. When I get back in front of a computer I will >> work it up. >> >> Sent from my iPhone > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 david.ponzone at gmail.com Thu Jun 3 10:03:22 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 3 Jun 2010 19:03:22 +0200 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <59C637855C5146D28EDE676D7A81405D@MOBILEE1705> References: <1275569298280-5134985.post@n2.nabble.com> <4535EF21-F837-456C-A167-82A8D914DC05@freeswitch.org> <59C637855C5146D28EDE676D7A81405D@MOBILEE1705> Message-ID: they have deep, large pockets :) David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 03/06/2010 ? 18:33, Madovsky a ?crit : > didn't know FS team were a pro apple ;) > hard to put an iPad in your pocket :D > > ----- Original Message ----- > From: "Brian West" > To: > Sent: Thursday, June 03, 2010 11:56 AM > Subject: Re: [Freeswitch-users] Help get freeswitch.com back > > >> Its time for an iPad. >> >> /b >> >> On Jun 3, 2010, at 10:51 AM, Michael S Collins wrote: >> >>> Definitely a good idea. When I get back in front of a computer I >>> will >>> work it up. >>> >>> Sent from my iPhone >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100603/26e9eb08/attachment.html From brian at freeswitch.org Thu Jun 3 10:12:12 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Jun 2010 12:12:12 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: <1275569298280-5134985.post@n2.nabble.com> <4535EF21-F837-456C-A167-82A8D914DC05@freeswitch.org> <59C637855C5146D28EDE676D7A81405D@MOBILEE1705> Message-ID: <05C3CFE7-EED0-4476-8754-20E23E0F0A67@freeswitch.org> Don't ya wish? :P /b On Jun 3, 2010, at 12:03 PM, David Ponzone wrote: > they have deep, large pockets :) > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/a8d0276a/attachment.html From david.ponzone at gmail.com Thu Jun 3 10:23:48 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 3 Jun 2010 19:23:48 +0200 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <05C3CFE7-EED0-4476-8754-20E23E0F0A67@freeswitch.org> References: <1275569298280-5134985.post@n2.nabble.com> <4535EF21-F837-456C-A167-82A8D914DC05@freeswitch.org> <59C637855C5146D28EDE676D7A81405D@MOBILEE1705> <05C3CFE7-EED0-4476-8754-20E23E0F0A67@freeswitch.org> Message-ID: <776139F9-FD92-4B26-A62E-A4E033D337D5@gmail.com> Sure we do :) I think I've overheard you saying you had 2 iPads. You really don't need that, I think you should spare one to Michael :) David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 03/06/2010 ? 19:12, Brian West a ?crit : > Don't ya wish? :P > > /b > > On Jun 3, 2010, at 12:03 PM, David Ponzone wrote: > >> they have deep, large pockets :) >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/53e085e0/attachment-0001.html From infos at madovsky.org Thu Jun 3 10:32:06 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 3 Jun 2010 13:32:06 -0400 Subject: [Freeswitch-users] Help get freeswitch.com back References: <1275569298280-5134985.post@n2.nabble.com><4535EF21-F837-456C-A167-82A8D914DC05@freeswitch.org><59C637855C5146D28EDE676D7A81405D@MOBILEE1705> Message-ID: <2ED20EDBB44244ABA9201427F8E38AFD@MOBILEE1705> Haaa, hope with no hooole ! :D ----- Original Message ----- From: David Ponzone To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 03, 2010 1:03 PM Subject: Re: [Freeswitch-users] Help get freeswitch.com back they have deep, large pockets :) David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 03/06/2010 ? 18:33, Madovsky a ?crit : didn't know FS team were a pro apple ;) hard to put an iPad in your pocket :D ----- Original Message ----- From: "Brian West" To: Sent: Thursday, June 03, 2010 11:56 AM Subject: Re: [Freeswitch-users] Help get freeswitch.com back Its time for an iPad. /b On Jun 3, 2010, at 10:51 AM, Michael S Collins wrote: Definitely a good idea. When I get back in front of a computer I will work it up. Sent from my iPhone _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/b422b167/attachment.html From devel at thom.fr.eu.org Thu Jun 3 10:36:49 2010 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Thu, 03 Jun 2010 19:36:49 +0200 Subject: [Freeswitch-users] Help debugging INVITE with provider Message-ID: Hello, I'm having trouble to send/receive calls with one of my providers. Freeswitch says the reply sent by provider has the session information incorrect, then the provider replies the calle is busy (which of course is not). The sip trace follows. Anybody can help ? Fran?ois send 1413 bytes to udp/[MyProviderIp]:5060 at 15:27:29.696419: ------------------------------------------------------------------------ INVITE sip:SomeAccount at provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 642 X-FS-Support: update_display Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1275553793 1275553794 IN IP4 MyFSIpHere s=FreeSWITCH c=IN IP4 MyFSIpHere t=0 0 m=audio 25056 RTP/AVP 124 123 122 121 3 99 9 8 0 101 13 a=rtpmap:124 G726-16/8000 a=rtpmap:123 G726-24/8000 a=rtpmap:122 G726-32/8000 a=rtpmap:121 G726-40/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 19186 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 H263-2000/90000 a=rtpmap:99 H264/90000 recv 420 bytes from udp/[MyProviderIp]:5060 at 15:27:29.770669: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 1030 bytes from udp/[MyProviderIp]:5060 at 15:27:29.774935: ------------------------------------------------------------------------ INVITE sip:gw+12voip.com at MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: Contact: sip:0000000000 at MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp CSeq: 1 INVITE User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 453 v=0 o=FrancoisLegal 1275578849 1275578849 IN IP4 80.239.235.106 s=SIP Call c=IN IP4 80.239.235.106 t=0 0 m=audio 11180 RTP/AVP 2 3 9 8 0 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 11598 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 a=rtpmap:99 a=ptime:20 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:29.775380: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 552 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866610: ------------------------------------------------------------------------ ACK sip:gw+12voip.com at MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK3fecf431e27c437190b2844df71b49b0 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:0000000000 at MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 423 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866859: ------------------------------------------------------------------------ SIP/2.0 486 Busy here Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK freeswitch at tls-srv-01> From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ send 320 bytes to udp/[MyProviderIp]:5060 at 15:27:29.867043: ------------------------------------------------------------------------ ACK sip:SomeAccount at provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 ACK Content-Length: 0 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:30.276655: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 498 bytes from udp/[MyProviderIp]:5060 at 15:27:30.341335: ------------------------------------------------------------------------ ACK sip:77.72.169.134:5060 SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:77.72.169.134:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/cf82aaf4/attachment-0001.html From jan.berger at video24.no Thu Jun 3 10:42:14 2010 From: jan.berger at video24.no (Jan Berger) Date: Thu, 3 Jun 2010 19:42:14 +0200 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <776139F9-FD92-4B26-A62E-A4E033D337D5@gmail.com> References: <1275569298280-5134985.post@n2.nabble.com> <4535EF21-F837-456C-A167-82A8D914DC05@freeswitch.org> <59C637855C5146D28EDE676D7A81405D@MOBILEE1705> <05C3CFE7-EED0-4476-8754-20E23E0F0A67@freeswitch.org> <776139F9-FD92-4B26-A62E-A4E033D337D5@gmail.com> Message-ID: <5278362B7972442A860BDB521AF88DF4@dell9400> He would probably by a 3rd for that purpose ? but Michael would still not have one :-) My wife have a stack of them ? apple stuff ? what amuses me most is that she need to borrow my old Windows machine all the time :) _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone Sent: 3. juni 2010 19:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Help get freeswitch.com back Sure we do :) I think I've overheard you saying you had 2 iPads. You really don't need that, I think you should spare one to Michael :) David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 03/06/2010 ? 19:12, Brian West a ?crit : Don't ya wish? :P /b On Jun 3, 2010, at 12:03 PM, David Ponzone wrote: they have deep, large pockets :) David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/e9b2213e/attachment.html From david.ponzone at gmail.com Thu Jun 3 10:53:19 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 3 Jun 2010 19:53:19 +0200 Subject: [Freeswitch-users] Help debugging INVITE with provider In-Reply-To: References: Message-ID: <07314285-F095-4946-B041-D50CD875326D@gmail.com> I would really recommend you resend the traces with the full headers (change the phone numbers and IPs if you need to, but replace them by variables so that we know when the same IP or number is present in various fields/headers). David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 03/06/2010 ? 19:36, Fran?ois Legal a ?crit : > Hello, > > > I'm having trouble to send/receive calls with one of my providers. > > Freeswitch says the reply sent by provider has the session > information incorrect, then the provider replies the calle is busy > (which of course is not). > > > The sip trace follows. > > > Anybody can help ? > > > Fran?ois > > > send 1413 bytes to udp/[MyProviderIp]:5060 at 15:27:29.696419: > > ------------------------------------------------------------------------ > INVITE sip:SomeAccount at provider.com SIP/2.0 > Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK > Max-Forwards: 70 > From: "" ;tag=2SU9FHayaSXcr > To: > Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e > CSeq: 131671088 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 642 > X-FS-Support: update_display > Remote-Party-ID: ;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1275553793 1275553794 IN IP4 MyFSIpHere > s=FreeSWITCH > c=IN IP4 MyFSIpHere > t=0 0 > m=audio 25056 RTP/AVP 124 123 122 121 3 99 9 8 0 101 13 > a=rtpmap:124 G726-16/8000 > a=rtpmap:123 G726-24/8000 > a=rtpmap:122 G726-32/8000 > a=rtpmap:121 G726-40/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:99 SPEEX/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > m=video 19186 RTP/AVP 31 34 115 121 99 > a=rtpmap:31 H261/90000 > a=rtpmap:34 H263/90000 > a=rtpmap:115 H263-1998/90000 > a=rtpmap:121 H263-2000/90000 > a=rtpmap:99 H264/90000 > > recv 420 bytes from udp/[MyProviderIp]:5060 at 15:27:29.770669: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK > From: "" ;tag=2SU9FHayaSXcr > To: > Contact: sip:MyProviderIp:5060 > Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e > CSeq: 131671088 INVITE > Server: (Very nice Sip Registrar/Proxy Server) > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 1030 bytes from udp/[MyProviderIp]:5060 at 15:27:29.774935: > > ------------------------------------------------------------------------ > INVITE sip:gw+12voip.com at MyFSIpHere:5060;transport=udp;gw=12voip.com > SIP/2.0 > Via: SIP/2.0/UDP MyProviderIp: > 5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 > From: "0000000000" ;tag=4e0113ac4c050c86c51a2 > To: > Contact: sip:0000000000 at MyProviderIp:5060 > Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp > CSeq: 1 INVITE > User-Agent: (Very nice Sip Registrar/Proxy Server) > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE > Content-Type: application/sdp > Content-Length: 453 > > v=0 > o=FrancoisLegal 1275578849 1275578849 IN IP4 80.239.235.106 > s=SIP Call > c=IN IP4 80.239.235.106 > t=0 0 > m=audio 11180 RTP/AVP 2 3 9 8 0 101 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > m=video 11598 RTP/AVP 31 34 115 121 99 > a=rtpmap:31 H261/90000 > a=rtpmap:34 H263/90000 > a=rtpmap:115 H263-1998/90000 > a=rtpmap:121 > a=rtpmap:99 > a=ptime:20 > > ------------------------------------------------------------------------ > send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:29.775380: > > ------------------------------------------------------------------------ > SIP/2.0 400 Bad Session Description > Via: SIP/2.0/UDP MyProviderIp: > 5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 > From: "0000000000" ;tag=4e0113ac4c050c86c51a2 > To: ;tag=32m2HcU171KZK > Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp > CSeq: 1 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, refer > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 552 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866610: > > ------------------------------------------------------------------------ > ACK sip:gw+12voip.com at MyFSIpHere:5060;transport=udp;gw=12voip.com > SIP/2.0 > Via: SIP/2.0/UDP MyProviderIp: > 5060;branch=z9hG4bK3fecf431e27c437190b2844df71b49b0 > From: "0000000000" ;tag=4e0113ac4c050c86c51a2 > To: ;tag=32m2HcU171KZK > Contact: sip:0000000000 at MyProviderIp:5060 > Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp > CSeq: 1 ACK > Server: (Very nice Sip Registrar/Proxy Server) > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 423 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866859: > > ------------------------------------------------------------------------ > SIP/2.0 486 Busy here > Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK > freeswitch at tls-srv-01> From: "" ;tag=2SU9FHayaSXcr > To: > Contact: sip:MyProviderIp:5060 > Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e > CSeq: 131671088 INVITE > Server: (Very nice Sip Registrar/Proxy Server) > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 320 bytes to udp/[MyProviderIp]:5060 at 15:27:29.867043: > > ------------------------------------------------------------------------ > ACK sip:SomeAccount at provider.com SIP/2.0 > Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK > Max-Forwards: 70 > From: "" ;tag=2SU9FHayaSXcr > To: > Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e > CSeq: 131671088 ACK > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:30.276655: > > ------------------------------------------------------------------------ > SIP/2.0 400 Bad Session Description > Via: SIP/2.0/UDP MyProviderIp: > 5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 > From: "0000000000" ;tag=4e0113ac4c050c86c51a2 > To: ;tag=32m2HcU171KZK > Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp > CSeq: 1 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, refer > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 498 bytes from udp/[MyProviderIp]:5060 at 15:27:30.341335: > > ------------------------------------------------------------------------ > ACK sip:77.72.169.134:5060 SIP/2.0 > Via: SIP/2.0/UDP MyProviderIp: > 5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 > From: "0000000000" ;tag=4e0113ac4c050c86c51a2 > To: ;tag=32m2HcU171KZK > Contact: sip:77.72.169.134:5060 > Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp > CSeq: 1 ACK > Server: (Very nice Sip Registrar/Proxy Server) > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE > Content-Length: 0 > > > ------------------------------------------------------------------------ > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/7448f94f/attachment-0001.html From brian at freeswitch.org Thu Jun 3 11:10:02 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Jun 2010 13:10:02 -0500 Subject: [Freeswitch-users] Help debugging INVITE with provider In-Reply-To: <07314285-F095-4946-B041-D50CD875326D@gmail.com> References: <07314285-F095-4946-B041-D50CD875326D@gmail.com> Message-ID: <75EEB095-BE8E-4F44-85A7-D9E3FE7203FD@freeswitch.org> Get a pcap and email it to me off list... /b On Jun 3, 2010, at 12:53 PM, David Ponzone wrote: > I would really recommend you resend the traces with the full headers (change the phone numbers and IPs if you need to, but replace them by variables so that we know when the same IP or number is present in various fields/headers). > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/075f964b/attachment.html From infos at madovsky.org Thu Jun 3 11:16:04 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 3 Jun 2010 14:16:04 -0400 Subject: [Freeswitch-users] check status of a number Message-ID: <9833524A0669475FA86895E58A23CF16@MOBILEE1705> Hi, I'd like to use fs_cli to check if the callee number exists and get the status result. Is anyone already did that ? Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/549f0034/attachment.html From brian at freeswitch.org Thu Jun 3 11:22:21 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Jun 2010 13:22:21 -0500 Subject: [Freeswitch-users] check status of a number In-Reply-To: <9833524A0669475FA86895E58A23CF16@MOBILEE1705> References: <9833524A0669475FA86895E58A23CF16@MOBILEE1705> Message-ID: Use the loopback endpoint with originate to do this. /b On Jun 3, 2010, at 1:16 PM, Madovsky wrote: > Hi, > > I'd like to use fs_cli to check if the callee number exists and get the status result. > > Is anyone already did that ? > > Thanks > > Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/03f41d1a/attachment.html From mgende at gendesign.com Thu Jun 3 11:25:15 2010 From: mgende at gendesign.com (Michael Gende) Date: Thu, 3 Jun 2010 13:25:15 -0500 Subject: [Freeswitch-users] Out-Going Call Transfer Question In-Reply-To: References: Message-ID: On Wed, Jun 2, 2010 at 1:42 PM, Michael Collins wrote: > Mike, > > Maybe you could pastebin a log of this behavior. If at all relevant, the output on the console reads thus when I try to transfer a completed, out-going call to another extension: I just called to my cell phone from FS. Then, having answered it, transfed the call from my desk to my colleague at ext 1012. The transfer hangs up the call on both ends. Console output follows: 2010-6-03 12:51:42:950492 [ERR] Switch_ivr_originate.c: 1494 Cannot Create outoing channel of type [user] Cause: [SUBSCRIBER ABSENT] 2010-6-03 12:51:42:950492 [INFO] mode_dtools.c: 2091 Originate Failed. Cause: SUBSCRIBER_ABSENT 2010-06-03 12:51:43.960389 [WARNING] mod_voicemail.c:2931 Can't Find User [1012 at wan.ip.address] > Also, is this a plain FS install with the default dialplan? If not then > include your relevant dialplan entries. Our install is pretty well documented in the following: http://wiki.freeswitch.org/wiki/Multi_home_tutorial The setup described here is what we have. > Lastly, are you on the most recent hit HEAD version of FS? > Nope. This is FS on PFSense, the Beta 0.9.7.26 version. No other issues, however, despite the Beta moniker. > > Thanks, > MC > Thanks for Responding. Need more? Let me know. Mike G. > > On Wed, Jun 2, 2010 at 8:28 AM, Michael Gende wrote: > >> Hello, >> >> I wonder if someone could direct me to some documentation (I'm sure it >> exists) on how to undertake something. >> >> We've been happily using FS for some time now. Our is set-up emulates a >> pretty standard key system. >> >> To the point of my post: I can, for incoming calls, easily and >> successfully transfer them to conference rooms and other registered >> extensions. >> >> However, if I call out to another party, I can only transfer the call to a >> conference room, not another extension. >> >> I suspect this has to do with FS not knowing - or me not correctly telling >> it how - to handle the "out going" case. I.e, which "end" of the call to >> transfer. >> >> Sometimes its handy to make a call and transfer the callee to some other >> extension. Any suggested reading? >> >> Thanks, >> >> Mike G. >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100603/69253180/attachment.html From infos at madovsky.org Thu Jun 3 11:50:24 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 3 Jun 2010 14:50:24 -0400 Subject: [Freeswitch-users] check status of a number References: <9833524A0669475FA86895E58A23CF16@MOBILEE1705> Message-ID: <607B5A3B3B484885A231E055D68D544B@MOBILEE1705> ok thanks F ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 03, 2010 2:22 PM Subject: Re: [Freeswitch-users] check status of a number Use the loopback endpoint with originate to do this. /b On Jun 3, 2010, at 1:16 PM, Madovsky wrote: Hi, I'd like to use fs_cli to check if the callee number exists and get the status result. Is anyone already did that ? Thanks Franck ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/cd8d044c/attachment-0001.html From mattdfong at gmail.com Thu Jun 3 12:03:12 2010 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 3 Jun 2010 12:03:12 -0700 Subject: [Freeswitch-users] Spikes in Load on Ubuntu Message-ID: I'm running 1.0.6 and I noticed that right after upgrading from 15135 that the load (15 min avg) on my system seems to have increased about double running the same amount of traffic. I've been running top all day for the past few days and I noticed that for the most part I always have 75% idle cpu, but maybe once ever 45 minutes or an hour there is an spike the 1 minute load averages (maybe up to 9). This is only a dual core processor so I believe 9 usually indicates that are some processes that are waiting. The idle % does not change, and I have free memory, but for whatever reason the linux load algorithm spikes (maybe very high for like 1-5 seconds). At first I thought this might be related to a tickless timer, so I disable it from the kernel, but today I am still getting the same spikes. So my question is, are these spikes something to worry about, or should I just monitor the % of idle cpu available on my box for guesstimating how much further I can push this system? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/cd511fee/attachment.html From asannucci at gmail.com Thu Jun 3 12:02:56 2010 From: asannucci at gmail.com (bakko) Date: Thu, 3 Jun 2010 21:02:56 +0200 Subject: [Freeswitch-users] Freeswicth nms In-Reply-To: <1275579239.1789.7.camel@lenovo400> References: <895033401-1275521270-cardhu_decombobulator_blackberry.rim.net-328548762-@bda057.bisx.prodap.on.blackberry> <1275579239.1789.7.camel@lenovo400> Message-ID: I'm using monit... :) From infos at madovsky.org Thu Jun 3 12:15:22 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 3 Jun 2010 15:15:22 -0400 Subject: [Freeswitch-users] check status of a number References: <9833524A0669475FA86895E58A23CF16@MOBILEE1705> Message-ID: I tried originate loopback/numberToCheck localNumber but it says didn't match anything in dialplan should I use special extension in dialplan ? Thanks ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 03, 2010 2:22 PM Subject: Re: [Freeswitch-users] check status of a number Use the loopback endpoint with originate to do this. /b On Jun 3, 2010, at 1:16 PM, Madovsky wrote: Hi, I'd like to use fs_cli to check if the callee number exists and get the status result. Is anyone already did that ? Thanks Franck ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/f3c84266/attachment.html From infos at madovsky.org Thu Jun 3 12:15:50 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 3 Jun 2010 15:15:50 -0400 Subject: [Freeswitch-users] Freeswicth nms References: <895033401-1275521270-cardhu_decombobulator_blackberry.rim.net-328548762-@bda057.bisx.prodap.on.blackberry><1275579239.1789.7.camel@lenovo400> Message-ID: <86ACEC07BA3045CB9EAE360A3A8F4E07@MOBILEE1705> also ganglia... ----- Original Message ----- From: "bakko" To: Sent: Thursday, June 03, 2010 3:02 PM Subject: Re: [Freeswitch-users] Freeswicth nms > I'm using monit... > > :) > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brian at freeswitch.org Thu Jun 3 12:19:41 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Jun 2010 14:19:41 -0500 Subject: [Freeswitch-users] check status of a number In-Reply-To: References: <9833524A0669475FA86895E58A23CF16@MOBILEE1705> Message-ID: Um isn't this exactly what you wanted to test? /b On Jun 3, 2010, at 2:15 PM, Madovsky wrote: > but it says didn't match anything in dialplan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/d523add1/attachment.html From brian at freeswitch.org Thu Jun 3 12:21:53 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Jun 2010 14:21:53 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] OSTAG - Open Source Telephony Action Group In-Reply-To: References: Message-ID: I would like to thank everyone that has donated to this cause. The money has been collected for the Open Source Telephony Advancement Group (I think Collins had a brain flub up). Thanks, /b On Jun 1, 2010, at 2:20 PM, Michael Collins wrote: > Good news! > > The Open Source Telephony Action Group (OSTAG) is almost ready to become an official 501(c)3 non-profit organization. We need to raise $800 in order to file the necessary paperwork. We have set up a PayPal account where we can all send our donations: > > donations at ostag.org > > Please assist us in moving this important organization forward. OSTAG is dedicated to the advancement of open source telecommuncations software that improves the lives of people all over the world. This is a most worthy cause! Please help now and in the future. > > -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/d622043a/attachment.html From infos at madovsky.org Thu Jun 3 12:32:33 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 3 Jun 2010 15:32:33 -0400 Subject: [Freeswitch-users] check status of a number References: <9833524A0669475FA86895E58A23CF16@MOBILEE1705> Message-ID: I'm not sure ... only want to be sure that the number dialed (landline or mobile) respond to call (let's say one ring) and get the result status (SIP status ?) before delete the command in fs_cli F ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 03, 2010 3:19 PM Subject: Re: [Freeswitch-users] check status of a number Um isn't this exactly what you wanted to test? /b On Jun 3, 2010, at 2:15 PM, Madovsky wrote: but it says didn't match anything in dialplan ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/ba4ebca0/attachment.html From mitch.capper at gmail.com Thu Jun 3 12:42:32 2010 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 3 Jun 2010 15:42:32 -0400 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <5278362B7972442A860BDB521AF88DF4@dell9400> References: <1275569298280-5134985.post@n2.nabble.com> <4535EF21-F837-456C-A167-82A8D914DC05@freeswitch.org> <59C637855C5146D28EDE676D7A81405D@MOBILEE1705> <05C3CFE7-EED0-4476-8754-20E23E0F0A67@freeswitch.org> <776139F9-FD92-4B26-A62E-A4E033D337D5@gmail.com> <5278362B7972442A860BDB521AF88DF4@dell9400> Message-ID: So looking at the @gial.com it also seemed very close to our favorite @ gmail.com, sure enough a google turned up this guy has registered many other domains under that email address (@gmail) of course I am not sure if you actually want to make contact but it may give a better chance (Then again an out of contact registrant is a lot easier to deal with from a dispute point of view). ~Mitch On Thu, Jun 3, 2010 at 1:42 PM, Jan Berger wrote: > He would probably by a 3rd for that purpose ? but Michael would still not > have one J > > > > My wife have a stack of them ? apple stuff ? what amuses me most is that > she need to borrow my old Windows machine all the time :) > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Ponzone > *Sent:* 3. juni 2010 19:24 > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Help get freeswitch.com back > > > > Sure we do :) > > > > I think I've overheard you saying you had 2 iPads. > > You really don't need that, I think you should spare one to Michael :) > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > *www.ipeva.fr* - *www.ipeva-studio.com* > > > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message > s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de > ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.* > > * * > > > > > > > > Le 03/06/2010 ? 19:12, Brian West a ?crit : > > > > Don't ya wish? :P > > > > /b > > > > On Jun 3, 2010, at 12:03 PM, David Ponzone wrote: > > > > they have deep, large pockets :) > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > *www.ipeva.fr* - *www.ipeva-studio.com* > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100603/e016238e/attachment-0001.html From brian at freeswitch.org Thu Jun 3 12:47:22 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Jun 2010 14:47:22 -0500 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: References: <1275569298280-5134985.post@n2.nabble.com> <4535EF21-F837-456C-A167-82A8D914DC05@freeswitch.org> <59C637855C5146D28EDE676D7A81405D@MOBILEE1705> <05C3CFE7-EED0-4476-8754-20E23E0F0A67@freeswitch.org> <776139F9-FD92-4B26-A62E-A4E033D337D5@gmail.com> <5278362B7972442A860BDB521AF88DF4@dell9400> Message-ID: <7E2ADFA7-71A1-4B3F-8D48-CAB324862978@freeswitch.org> I have emailed that one multiple times too... no response. /b On Jun 3, 2010, at 2:42 PM, Mitch Capper wrote: > So looking at the @gial.com it also seemed very close to our favorite @gmail.com, sure enough a google turned up this guy has registered many other domains under that email address (@gmail) of course I am not sure if you actually want to make contact but it may give a better chance (Then again an out of contact registrant is a lot easier to deal with from a dispute point of view). > > ~Mitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/6e1d4ce7/attachment.html From mattdfong at gmail.com Thu Jun 3 13:16:54 2010 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 3 Jun 2010 13:16:54 -0700 Subject: [Freeswitch-users] Help get freeswitch.com back In-Reply-To: <7E2ADFA7-71A1-4B3F-8D48-CAB324862978@freeswitch.org> References: <1275569298280-5134985.post@n2.nabble.com> <4535EF21-F837-456C-A167-82A8D914DC05@freeswitch.org> <59C637855C5146D28EDE676D7A81405D@MOBILEE1705> <05C3CFE7-EED0-4476-8754-20E23E0F0A67@freeswitch.org> <776139F9-FD92-4B26-A62E-A4E033D337D5@gmail.com> <5278362B7972442A860BDB521AF88DF4@dell9400> <7E2ADFA7-71A1-4B3F-8D48-CAB324862978@freeswitch.org> Message-ID: you might be able to use this http://www.adr.eu/arbitration_platform/fees.php Resolution provider for .com which has lower fees. They are listed as ICANN approved UDRP (.com, I think). Also if the contact information is wrong, you can file a request with the Registrar to force an update, if they end up not updating the information with real information, the domain can go thru a long process of finally being released. hope this helps.. --matt On Thu, Jun 3, 2010 at 12:47 PM, Brian West wrote: > I have emailed that one multiple times too... no response. > > /b > > On Jun 3, 2010, at 2:42 PM, Mitch Capper wrote: > > So looking at the @gial.com it also seemed very close to our favorite @ > gmail.com, sure enough a google turned up this guy has registered many > other domains under that email address (@gmail) of course I am not sure if > you actually want to make contact but it may give a better chance (Then > again an out of contact registrant is a lot easier to deal with from a > dispute point of view). > > ~Mitch > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/e8934cb5/attachment.html From anthony.minessale at gmail.com Thu Jun 3 13:24:02 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Jun 2010 15:24:02 -0500 Subject: [Freeswitch-users] Spikes in Load on Ubuntu In-Reply-To: References: Message-ID: I'd start by not using ubuntu in production. If you insist: You need to run your kernel at 1000hz and not tickless. We recommend CentOS because it's been the most stable and reliable distro we have found. On Thu, Jun 3, 2010 at 2:03 PM, Matthew Fong wrote: > I'm running 1.0.6 and I noticed that right after upgrading from 15135 that > the load (15 min avg) on my system seems to have increased about double > running the same amount of traffic. I've been running top all day for the > past few days and I noticed that for the most part I always have 75% idle > cpu, but maybe once ever 45 minutes or an hour there is an spike the 1 > minute load averages (maybe up to 9). This is only a dual core processor so > I believe 9 usually indicates that are some processes that are waiting. The > idle % does not change, and I have free memory, but for whatever reason the > linux load algorithm spikes (maybe very high for like 1-5 seconds). > > At first I thought this might be related to a tickless timer, so I disable > it from the kernel, but today I am still getting the same spikes. So my > question is, are these spikes something to worry about, or should I just > monitor the % of idle cpu available on my box for guesstimating how much > further I can push this system? Thanks. > > --matt > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100603/64a3648e/attachment.html From djbinter at gmail.com Thu Jun 3 14:45:58 2010 From: djbinter at gmail.com (DJB INTERNATIONAL) Date: Thu, 3 Jun 2010 14:45:58 -0700 Subject: [Freeswitch-users] Question regarding codec param Message-ID: I would like to know whether when I set disable-transcoding=true, would it be the same as setting inherit_codec = true? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/d34a32b1/attachment.html From brian at freeswitch.org Thu Jun 3 14:51:49 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Jun 2010 16:51:49 -0500 Subject: [Freeswitch-users] Question regarding codec param In-Reply-To: References: Message-ID: <959E3DC1-8B69-493D-B4C3-CDC009D6F0B9@freeswitch.org> NO. It wouldn't. See disable transcoding causes the negotiated codec from the A leg to be the only codec offered on the B-Leg. While inherit_codec lets you enable late negotiation and the B-Leg decides then what ever that was will be what is 200OK'ed to the A-Leg Its different but the results are the same. /b On Jun 3, 2010, at 4:45 PM, DJB INTERNATIONAL wrote: > I would like to know whether when I set disable-transcoding=true, would it be the same as setting inherit_codec = true? > > Thank you. From codeghar at gmail.com Thu Jun 3 19:43:58 2010 From: codeghar at gmail.com (Code Ghar) Date: Thu, 3 Jun 2010 21:43:58 -0500 Subject: [Freeswitch-users] Freeswicth nms In-Reply-To: <1111639352-1275526834-cardhu_decombobulator_blackberry.rim.net-2040530345-@bda2058.bisx.prodap.on.blackberry> References: <1111639352-1275526834-cardhu_decombobulator_blackberry.rim.net-2040530345-@bda2058.bisx.prodap.on.blackberry> Message-ID: I would also recommend Nagios. Although we are not doing anything funky, it has saved our skin multiple times. It is fairly easy to use with lots of plugins. I haven't used it with FS, though. Whatever you use please try to share your config, etc. On Wed, Jun 2, 2010 at 8:00 PM, wrote: > Budi, > Try Nagios for network monitoring > > Regards > > Suvir > ------Original Message------ > From: Budi wibowo > Sender: freeswitch-users-bounces at lists.freeswitch.org > To: freeswitch-users at lists.freeswitch.org > ReplyTo: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Freeswicth nms > Sent: Jun 3, 2010 06:26 > > Hi all > I'm looking for nms that can do minimum alarm,fault, performance and > statistic. Alarm could be sent via sms or email. > I need the such solution due I have project to involve more than 10 fs box. > Both free and commercial solution are ok > > Regards > Budi > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Sent via BlackBerry? from AIS > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/e239b81f/attachment.html From infos at madovsky.org Thu Jun 3 19:52:32 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 3 Jun 2010 22:52:32 -0400 Subject: [Freeswitch-users] check status of a number References: <9833524A0669475FA86895E58A23CF16@MOBILEE1705> Message-ID: <9096DDFDFF1C4708AD290F88E4E18683@MOBILEE1705> Sorry to ask help again, I have no idea of how to use loopback with remote endpoint and originate. the link http://wiki.freeswitch.org/wiki/Mod_loopback shows only the path of local extension without any originate command example. I googled but no success. Thanks F ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 03, 2010 2:22 PM Subject: Re: [Freeswitch-users] check status of a number Use the loopback endpoint with originate to do this. /b On Jun 3, 2010, at 1:16 PM, Madovsky wrote: Hi, I'd like to use fs_cli to check if the callee number exists and get the status result. Is anyone already did that ? Thanks Franck ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/dd4e1b8a/attachment-0001.html From codeghar at gmail.com Thu Jun 3 20:00:54 2010 From: codeghar at gmail.com (Code Ghar) Date: Thu, 3 Jun 2010 22:00:54 -0500 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: References: Message-ID: Hi David Have you (or someone here) worked with Kamailio? If it can be used to act as stateful proxy, it could work this way: Kamailio (10.10.1.1) takes the packet, determines if it is part of a certain call (maybe look at Call ID, etc.) -- or if it's a new call to pick one server -- and sends the packet to FS1 (10.10.1.2); FS1 responds with SDP containing its own IP for media; Kamilio modifies the headers (especially Contact to set its own IP instead of FS IP) and sends to carrier. This way we can have multiple FS media servers running behind one signaling IP. Since all FS servers have their own external IPs reachable from the outside network, their IPs in effect appear as media IPs to carriers and they send signaling to Kamailio but media directly to the FS server actually handling the call. Yes, it's not a pure FS solution but best tool for the job, right? Of course, I am assuming (having never worked with Kamailio) that it can handle this kind of architecture. I think it should be able to handle signaling for lots and lots of simultaneous calls because it is not handling media at all, freeing up resources for signaling only. Any thoughts? On Fri, May 28, 2010 at 6:31 AM, David Ponzone wrote: > Code, > > you're totally right. > In this model (FS), the media server will also be in the SIP Path. > That's why I answered in the first place that this was not achievable with > FS, because your idea was more a Kamaillo/RTPProxy setup, where the > mediaserver only does RTP with the endpoints, and is not in the SIP path at > all: > > inbound <--------SIP------ SIP Server/Proxy ------------SIP-------> Carrier > | > <---------RTP------ MediaServer--------RTP---------------> > > > Verizon Business (in Europe at least) has a such infrastrucutre: OpenSER > for the SIP part, and Nortel GWs for the RTP. > This way, they just give me the IPs of their OpenSER servers, and they can > deploy as many media servers as they need without telling us (of course, we > dont filter that). > > I don't know how this is implemented in Kamaillo/OpenSER but perhaps, for a > nice bounty, that would be something possible in FS. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 28/05/2010 ? 05:34, Code Ghar a ?crit : > > Hi Vitalie > > Thanks for providing the link and details. If I understood correctly, the > chain of signaling would be Inbound -> FSSIP -> FSRTP -> Outbound (using > names and terms in my original question), while the chain of media would be > Inbound -> FSRTP -> Outbound. This way we can have multiple servers handling > media and minimal servers handling signaling. > > Let me clarify a little more my motivation for asking this question in the > first place. I work with telecom carriers on a daily basis and have seen > many different architectures. The first biggest problem is how to load > balance SIP traffic when you are receiving calls, if one server is not > enough. The second biggest problem is handling all RTP, including > transcoding. With this architecture, one or two IPs for signaling can be > handled by most carriers. So you can beef up your hardware for signaling and > depend less on your carrier's ability to load balance traffic for you. If > they can do round-robin or failover for two IPs, you are golden. And then > you can deploy multiple nodes to handle all RTP duties, without having to > concern your carrier about multiple servers and IPs. But there's one thing > still missing. Your outbound carrier still needs to allow traffic from > multiple IPs because now they are dealing with FSRTP instead of FSSIP. > > Is there such a solution possible using FS that all signaling, for both > inbound and outbound carriers, can be handled by a couple of FSSIP nodes > (depending on the amount of traffic, maybe a few more) while delegating > media responsibilities to multiple FSRTP nodes? In this situation, SIP IP is > always, say 10.10.10.1 or 10.10.10.2, but the SDP may use media IPs > 10.10.10.3, 10.10.10.4, 10.10.10.5, and so on. Almost all carriers I have > seen concern themselves only with which SIP IPs they should allow and don't > care how many or which media IPs they have to deal with. Any ideas on > minimizing signaling IPs while adding more media IPs as required? > > I have seen re-invite being used in production where you can just let your > inbound and outbound handle media between them on their own without it going > through your network. But there are circumstances where people might need to > pass media through their own network, chiefly to perform transcoding and > also to handle other interop issues. It is because of this use case, > combined with the need for minimizing signaling IPs, that the original > question was asked. > > > > > On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov wrote: > >> Hi Code, >> >> I have working example doing exactly what you've described. >> One signalling FS bridges incoming call to a set of media servers >> (depending on ip, but you can implement any routing logic including round >> robin) and then transfers media stream after bridging to that media server. >> >> You can achieve this on signalling FS by creating a Lua script that >> contains the following lines: >> >> media_server="my_media_X.mydomain.com"; --to be determined by routing >> logic >> forwarding_session = "sofia/external/"..called_number.."@"..media_server; >> session:setVariable("bypass_media_after_bridge", "true"); >> session:setVariable("hangup_after_bridge", "true"); >> session:execute("bridge",forwarding_session); >> >> The call will arrive to the selected media server successfully and media >> stream will start bypassing signalling FS after bridge. >> >> You can read the following thread, it describes how you can setup such >> configuration. >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html >> >> I think it will fit your needs. >> >> Regards, >> Vitalie >> >> >> 2010/5/27 Code Ghar >> >>> Is it possible -- and are there any case studies, practical experience, >>> etc -- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP) >>> handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ..., >>> FSRTPn) handle all media responsibilities? So when a call comes in, the SDP >>> contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would >>> request FSRTPx for media resources for each new call and add its IP and port >>> in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and >>> forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP >>> only deals with signaling. This way multiple servers could be deployed to >>> handle media responsibilities and only a handful would be required for >>> signaling. In future if there's a greater need for transcoding, etc. all you >>> need to do is deploy a media server and not have to add servers for >>> signaling. >>> >>> This idea came to me because I have come across two proprietary >>> applications that do it this way. They have a SIP component and a media >>> component. You can run both on the same physical machine or you can separate >>> them out into multiple machines. >>> >>> Another way for this could be to integrate FS as a media component to >>> another application's SIP component. A mix-and-match, so to speak. >>> >>> On the flip side, deploy FS as a SIP server and use media capabilities of >>> some other hardware or software application. For example, FS handles >>> signaling and use dedicated hardware for media. A good example of this is >>> illustrated (somewhat) by an image on Sangoma's website: >>> http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg. >>> Look at the "pooled transcoding". >>> >>> Is FS even designed to be this modular? If so, how can the aforementioned >>> scenario(s) be achieved? >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100603/4886011b/attachment.html From infos at madovsky.org Thu Jun 3 20:51:25 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 3 Jun 2010 23:51:25 -0400 Subject: [Freeswitch-users] check status of a number References: <9833524A0669475FA86895E58A23CF16@MOBILEE1705> Message-ID: <85E0E8E1683646568D9E1929D7CD1108@MOBILEE1705> Ok I succeed doing like this : /usr/local/freeswitch/bin/fs_cli -x "expand originate sofia/gateway/\${distributor(pstn_international_2)}/003434REMOTEEXT loopback/999LOCALEXT" and it gives a session: +OK d0c598b5-9291-47eb-a14e-7845763425f4 does +OK means the number always exists ? is there only one ring, hangup and clean the originate call ? Thanks F ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 03, 2010 2:22 PM Subject: Re: [Freeswitch-users] check status of a number Use the loopback endpoint with originate to do this. /b On Jun 3, 2010, at 1:16 PM, Madovsky wrote: Hi, I'd like to use fs_cli to check if the callee number exists and get the status result. Is anyone already did that ? Thanks Franck ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100603/715207e3/attachment-0001.html From david.ponzone at gmail.com Thu Jun 3 22:59:32 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Fri, 4 Jun 2010 07:59:32 +0200 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: References: Message-ID: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> It doesn't solve the issue that all the media servers will do signaling too, and will talk SIP with the carriers. So the carriers will need to allow all the media servers . The only clean solution to avoid that, I think, is to have signaling boxes allocating resources from media servers with another protocol than SIP. RTPproxy does that I think, but I am not sure how it works. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 04/06/2010 ? 05:00, Code Ghar a ?crit : > Hi David > > Have you (or someone here) worked with Kamailio? If it can be used > to act as stateful proxy, it could work this way: Kamailio > (10.10.1.1) takes the packet, determines if it is part of a certain > call (maybe look at Call ID, etc.) -- or if it's a new call to pick > one server -- and sends the packet to FS1 (10.10.1.2); FS1 responds > with SDP containing its own IP for media; Kamilio modifies the > headers (especially Contact to set its own IP instead of FS IP) and > sends to carrier. > > This way we can have multiple FS media servers running behind one > signaling IP. Since all FS servers have their own external IPs > reachable from the outside network, their IPs in effect appear as > media IPs to carriers and they send signaling to Kamailio but media > directly to the FS server actually handling the call. > > Yes, it's not a pure FS solution but best tool for the job, right? > Of course, I am assuming (having never worked with Kamailio) that it > can handle this kind of architecture. I think it should be able to > handle signaling for lots and lots of simultaneous calls because it > is not handling media at all, freeing up resources for signaling only. > > Any thoughts? > > > On Fri, May 28, 2010 at 6:31 AM, David Ponzone > wrote: > Code, > > you're totally right. > In this model (FS), the media server will also be in the SIP Path. > That's why I answered in the first place that this was not > achievable with FS, because your idea was more a Kamaillo/RTPProxy > setup, where the mediaserver only does RTP with the endpoints, and > is not in the SIP path at all: > > inbound <--------SIP------ SIP Server/Proxy ------------SIP-------> > Carrier > | > <---------RTP------ MediaServer-------- > RTP---------------> > > > Verizon Business (in Europe at least) has a such infrastrucutre: > OpenSER for the SIP part, and Nortel GWs for the RTP. > This way, they just give me the IPs of their OpenSER servers, and > they can deploy as many media servers as they need without telling > us (of course, we dont filter that). > > I don't know how this is implemented in Kamaillo/OpenSER but > perhaps, for a nice bounty, that would be something possible in FS. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 28/05/2010 ? 05:34, Code Ghar a ?crit : > >> Hi Vitalie >> >> Thanks for providing the link and details. If I understood >> correctly, the chain of signaling would be Inbound -> FSSIP -> >> FSRTP -> Outbound (using names and terms in my original question), >> while the chain of media would be Inbound -> FSRTP -> Outbound. >> This way we can have multiple servers handling media and minimal >> servers handling signaling. >> >> Let me clarify a little more my motivation for asking this question >> in the first place. I work with telecom carriers on a daily basis >> and have seen many different architectures. The first biggest >> problem is how to load balance SIP traffic when you are receiving >> calls, if one server is not enough. The second biggest problem is >> handling all RTP, including transcoding. With this architecture, >> one or two IPs for signaling can be handled by most carriers. So >> you can beef up your hardware for signaling and depend less on your >> carrier's ability to load balance traffic for you. If they can do >> round-robin or failover for two IPs, you are golden. And then you >> can deploy multiple nodes to handle all RTP duties, without having >> to concern your carrier about multiple servers and IPs. But there's >> one thing still missing. Your outbound carrier still needs to allow >> traffic from multiple IPs because now they are dealing with FSRTP >> instead of FSSIP. >> >> Is there such a solution possible using FS that all signaling, for >> both inbound and outbound carriers, can be handled by a couple of >> FSSIP nodes (depending on the amount of traffic, maybe a few more) >> while delegating media responsibilities to multiple FSRTP nodes? In >> this situation, SIP IP is always, say 10.10.10.1 or 10.10.10.2, but >> the SDP may use media IPs 10.10.10.3, 10.10.10.4, 10.10.10.5, and >> so on. Almost all carriers I have seen concern themselves only with >> which SIP IPs they should allow and don't care how many or which >> media IPs they have to deal with. Any ideas on minimizing signaling >> IPs while adding more media IPs as required? >> >> I have seen re-invite being used in production where you can just >> let your inbound and outbound handle media between them on their >> own without it going through your network. But there are >> circumstances where people might need to pass media through their >> own network, chiefly to perform transcoding and also to handle >> other interop issues. It is because of this use case, combined with >> the need for minimizing signaling IPs, that the original question >> was asked. >> >> >> >> >> On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov >> wrote: >> Hi Code, >> >> I have working example doing exactly what you've described. >> One signalling FS bridges incoming call to a set of media servers >> (depending on ip, but you can implement any routing logic including >> round robin) and then transfers media stream after bridging to that >> media server. >> >> You can achieve this on signalling FS by creating a Lua script that >> contains the following lines: >> >> media_server="my_media_X.mydomain.com"; --to be determined by >> routing logic >> forwarding_session = "sofia/ >> external/"..called_number.."@"..media_server; >> session:setVariable("bypass_media_after_bridge", "true"); >> session:setVariable("hangup_after_bridge", "true"); >> session:execute("bridge",forwarding_session); >> >> The call will arrive to the selected media server successfully and >> media stream will start bypassing signalling FS after bridge. >> >> You can read the following thread, it describes how you can setup >> such configuration. >> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html >> >> I think it will fit your needs. >> >> Regards, >> Vitalie >> >> >> 2010/5/27 Code Ghar >> Is it possible -- and are there any case studies, practical >> experience, etc -- on deploying FreeSWITCH (FS) in this >> architecture: one server (FSSIP) handles SIP signaling only, and >> multiple servers (FSRTP1, FSRTP2, ..., FSRTPn) handle all media >> responsibilities? So when a call comes in, the SDP contains IP of, >> say FSRTP1, as media handler. For this to work, FSSIP would request >> FSRTPx for media resources for each new call and add its IP and >> port in SDP. The media servers/gateways would play IVR, etc.; >> collect DTMF and forward as appropriate to FSSIP; perform >> transcoding; etc.; all while FSSIP only deals with signaling. This >> way multiple servers could be deployed to handle media >> responsibilities and only a handful would be required for >> signaling. In future if there's a greater need for transcoding, >> etc. all you need to do is deploy a media server and not have to >> add servers for signaling. >> >> This idea came to me because I have come across two proprietary >> applications that do it this way. They have a SIP component and a >> media component. You can run both on the same physical machine or >> you can separate them out into multiple machines. >> >> Another way for this could be to integrate FS as a media component >> to another application's SIP component. A mix-and-match, so to speak. >> >> On the flip side, deploy FS as a SIP server and use media >> capabilities of some other hardware or software application. For >> example, FS handles signaling and use dedicated hardware for media. >> A good example of this is illustrated (somewhat) by an image on >> Sangoma's website: http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg >> . Look at the "pooled transcoding". >> >> Is FS even designed to be this modular? If so, how can the >> aforementioned scenario(s) be achieved? >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/6c1e444b/attachment-0001.html From steveayre at gmail.com Fri Jun 4 00:38:25 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 4 Jun 2010 08:38:25 +0100 Subject: [Freeswitch-users] check status of a number In-Reply-To: <85E0E8E1683646568D9E1929D7CD1108@MOBILEE1705> References: <9833524A0669475FA86895E58A23CF16@MOBILEE1705> <85E0E8E1683646568D9E1929D7CD1108@MOBILEE1705> Message-ID: It means the originate command succeeded to create a call, which that UUID identifies. It doesn't tell you anything about the result of that call though - you'll need to watch for events for that UUID, or check its XML CDR to find out the callflow (to see if it matched an extension) or the hangup reason. -Steve On 4 June 2010 04:51, Madovsky wrote: > Ok I succeed doing like this : > > /usr/local/freeswitch/bin/fs_cli -x "expand originate > sofia/gateway/\${distributor(pstn_international_2)}/003434REMOTEEXT loopback/999LOCALEXT" > > and it gives a session: > > +OK d0c598b5-9291-47eb-a14e-7845763425f4 > > does +OK means the number always exists ? is there only one ring, hangup > and clean the originate call ? > > Thanks > > F > > ----- Original Message ----- > *From:* Brian West > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, June 03, 2010 2:22 PM > *Subject:* Re: [Freeswitch-users] check status of a number > > Use the loopback endpoint with originate to do this. > > /b > > On Jun 3, 2010, at 1:16 PM, Madovsky wrote: > > Hi, > > I'd like to use fs_cli to check if the callee number exists and get the > status result. > > Is anyone already did that ? > > Thanks > > Franck > > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100604/2b05034d/attachment.html From vfclists at gmail.com Fri Jun 4 00:55:00 2010 From: vfclists at gmail.com (Frank Church) Date: Fri, 4 Jun 2010 08:55:00 +0100 Subject: [Freeswitch-users] Can freeswitch work with ODBC for Microsoft Access files? Message-ID: I'd just like to know if any brave souls have tested Freeswitch with Microsoft ODBC for JET 4.0 using Microsoft Access or any of the other file formats it supports, like DBF and Paradox db files. Anyone? -- Frank Church ======================= http://devblog.brahmancreations.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/517326f4/attachment.html From devel at thom.fr.eu.org Fri Jun 4 00:57:54 2010 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Fri, 04 Jun 2010 09:57:54 +0200 Subject: [Freeswitch-users] Help debugging INVITE with provider In-Reply-To: <07314285-F095-4946-B041-D50CD875326D@gmail.com> References: <07314285-F095-4946-B041-D50CD875326D@gmail.com> Message-ID: <404ccbd42e100f414bab82b5bd65461e@thom.fr.eu.org> This is what I did (except for the last trace) : MyProviderIp is the IP FS tries to contact to reach SomeAccount at provider.com, and MyFSIpHere is the IP FS external profile is bound to. Maybe this lack some information : the "dialed number" is originate sofia/external/SomeAccount at provider.com which should at the end come back to my FS external profile through the gateway 12voip.com included in the external profile. Fran?ois On Thu, 3 Jun 2010 19:53:19 +0200, David Ponzone wrote: I would really recommend you resend the traces with the full headers (change the phone numbers and IPs if you need to, but replace them by variables so that we know when the same IP or number is present in various fields/headers). David Ponzone Direction Technique email: david.ponzone at ipeva.fr [1] tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com _Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. __IPEVA__ d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur._ Le 03/06/2010 ? 19:36, Fran?ois Legal a ?crit : Hello, I'm having trouble to send/receive calls with one of my providers. Freeswitch says the reply sent by provider has the session information incorrect, then the provider replies the calle is busy (which of course is not). The sip trace follows. Anybody can help ? Fran?ois send 1413 bytes to udp/[MyProviderIp]:5060 at 15:27:29.696419: ------------------------------------------------------------------------ INVITE sip:SomeAccount at provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 642 X-FS-Support: update_display Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1275553793 1275553794 IN IP4 MyFSIpHere s=FreeSWITCH c=IN IP4 MyFSIpHere t=0 0 m=audio 25056 RTP/AVP 124 123 122 121 3 99 9 8 0 101 13 a=rtpmap:124 G726-16/8000 a=rtpmap:123 G726-24/8000 a=rtpmap:122 G726-32/8000 a=rtpmap:121 G726-40/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 19186 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 H263-2000/90000 a=rtpmap:99 H264/90000 recv 420 bytes from udp/[MyProviderIp]:5060 at 15:27:29.770669: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 1030 bytes from udp/[MyProviderIp]:5060 at 15:27:29.774935: ------------------------------------------------------------------------ INVITE sip:gw+12voip.com at MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: Contact: sip:0000000000 at MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp CSeq: 1 INVITE User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 453 v=0 o=FrancoisLegal 1275578849 1275578849 IN IP4 80.239.235.106 s=SIP Call c=IN IP4 80.239.235.106 t=0 0 m=audio 11180 RTP/AVP 2 3 9 8 0 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 11598 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 a=rtpmap:99 a=ptime:20 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:29.775380: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 552 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866610: ------------------------------------------------------------------------ ACK sip:gw+12voip.com at MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK3fecf431e27c437190b2844df71b49b0 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:0000000000 at MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 423 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866859: ------------------------------------------------------------------------ SIP/2.0 486 Busy here Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK freeswitch at tls-srv-01> From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ send 320 bytes to udp/[MyProviderIp]:5060 at 15:27:29.867043: ------------------------------------------------------------------------ ACK sip:SomeAccount at provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 ACK Content-Length: 0 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:30.276655: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 498 bytes from udp/[MyProviderIp]:5060 at 15:27:30.341335: ------------------------------------------------------------------------ ACK sip:77.72.169.134:5060 SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:77.72.169.134:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c at MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Links: ------ [1] mailto:david.ponzone at ipeva.fr [2] mailto:FreeSWITCH-users at lists.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/1965e374/attachment-0001.html From christian.loeschenkohl at xpirio.com Fri Jun 4 02:14:10 2010 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Fri, 04 Jun 2010 11:14:10 +0200 Subject: [Freeswitch-users] outbound socket problem / fs_ivrd general protection, segfault Message-ID: <4C08C3E2.3020703@xpirio.com> hello list recently i have problems with outbound socket scripts (written in php) dialplan looks like this ... do something ... script1.php is executed script2.php fails (see syslog msg) syslog says fs_ivrd[11971] general protection ip:7f2f7b94bbf5 sp:7fffffffa148 error:0 in libpthread-2.7.so[7f2f7b942000+16000] or fs_ivrd[12037]: segfault at a006c6c7576 ip 7f2f7b94bbf5 sp 7fffffffa148 error 4 in libpthread-2.7.so[7f2f7b942000+16000] uname -a Linux vts02.vie.xpirio.net 2.6.26-2-amd64 #1 SMP Tue Mar 9 22:29:32 UTC 2010 x86_64 GNU/Linux version FreeSWITCH Version 1.0.6 (svn-exported) does anybody have similar problems? br -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From Prometheus001 at gmx.net Fri Jun 4 02:38:29 2010 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 04 Jun 2010 11:38:29 +0200 Subject: [Freeswitch-users] Sending NOTIFY message to a phone via XML-RPC Message-ID: <4C08C995.8060500@gmx.net> Hello, I try to send a NOTIFY message from my application to a registered phone via XML-RPC. This is the XML which is sent: freeswitch.api sendevent NOTIFY,profile=internal,event-string=check-sync;reboot=false,user=200,host=192.168.178.220,content-type=application/simple-message-summary However I receive an error message: . . . ERROR!. . . I also tried to send the data as a struct but FS complains about needing a string as parameters. What am I doing wrong? Anybody has a valid sample XML for a NOTIFY message? Best regards Peter From john_re at fastmail.us Fri Jun 4 03:32:43 2010 From: john_re at fastmail.us (giovanni_re) Date: Fri, 04 Jun 2010 03:32:43 -0700 Subject: [Freeswitch-users] BerkeleyTIP Join June Global Free SW HW Culture Mtgs via VOIP or in Berkeley Message-ID: <1275647563.9491.1378441381@webmail.messagingengine.com> You're invited to join in with the friendly people at the BerkeleyTIP global meeting - newbie to Ph.D. - everyone is invited. Get a headset & join using VOIP online, or come to Berkeley. 1st step: Join the mailing list: http://groups.google.com/group/BerkTIPGlobal Watch the videos. Discuss them on VOIP. 8 great videos/talks this month - see below. Starting off year 3 of BerkeleyTIP :) Join with us at the LOCATION TO BE DETERMINED at the University of California at Berkeley, or join from your home via VOIP, or send this email locally, create a local meeting, & join via VOIP: Tip: a wifi cafe is a great place to meet. :) JUNE 5 & 20 AT UCB MEETING LOCATIONS TO BE DETERMINED. PLEASE VIEW THE BTIP WEBSITE & MAILING LIST PAGES FOR THE LATEST DETIALS. http://sites.google.com/site/berkeleytip http://groups.google.com/group/BerkTIPGlobal BerkeleyTIP - Educational, Productive, Social For Learning about, Sharing, & Producing, All Free SW HW & Culture. TIP == Talks, Installfest, Project & Programming Party http://sites.google.com/site/berkeleytip ===== CONTENTS: 1) 2010 JUNE VIDEOS; 2) 2010 JUNE MEETING DAYS, TIMES, LOCATIONS; 3) LOCAL MEETING AT U. C. Berkeley LOCATION TO BE DETERMINED; 4) HOT TOPICS; 5) PLEASE RSVP PROBABILISTICALLY, THANKS :) ; 6) INSTALLFEST; SPECIAL: FREE 7th Annual BERKELEY WORLD MUSIC FESTIVAL June 5 Sat; 7) ARRIVING FIRST AT THE MEETING: MAKE A "BerkeleyTIP" SIGN; 8) IRC: #berkeleytip on irc.freenode.net; 9) VOIP FOR GLOBAL MEETING; 10) VOLUNTEERING, TO DOs; 11) MAILING LISTS: BerkeleyTIP-Global, LocalBerkeley, Announce; 12) ANYTHING I FORGOT TO MENTION?; 13) FOR FORWARDING ======================================================================= ===== 1) 2010 JUNE VIDEOS Super Computing for Business, Brian Modra, CLUG State of the Linux Union 2010, Jim Zemlin, Linux Foundation Journaled Soft-Updates, Dr. Kirk McKusick, BSDCan 2010 Scientific data visualization using Mayavi2, Gael Varoquaux, Python4ScienceUCB Bringing OLPC to children in Afghanistan, Carol Ruth Silver, OLPC-SF Text-to-Speech in Ubuntu with Kttsd Kmouth Festival, blip.tv Rabbi Rabbs, the UnixRabbi, leads a group of Unix geeks, Comedy, UUASC, BS"D, 2003 The Great Debate - Are We Alone?, Geoff Marcy and Dan Werthimer, SETI at UC Berkeley Thanks to all the speakers, organizations, & videographers. :) [Please alert the speakers that their talks are scheduled for June for BTIP (if you are with the group that recorded their talk), because I may not have time to do that. Thanks. :) ] URLs for video download & full details: http://sites.google.com/site/berkeleytip/talk-videos Download & watch these talks before the BTIP meetings. Discuss at the meeting. Email the mailing list, tell us what videos you'll watch & want to discuss: http://groups.google.com/group/BerkTIPGlobal Know any other video sources? - please email me. _Your_ group should video record & post online your meeting's talks! ===== 2) 2010 JUNE MEETING DAYS, TIMES, LOCATIONS In person meetings on 1st Saturday & 3rd Sunday, every month. June 5 & 20, 12N-3P USA-Pacific time, Saturday, Sunday Online only meeting using VOIP: June 14 & 29, 5-6P USA-Pacific time, Monday, Tuesday Mark your calendars. 5 Sat 12N-3P PDST = 3-6P EDST = 19-22 UTC 14 Mon 5-6P PDST = 8-9P EDST = 0- 1 UTC Tues 15 20 Sun 12N-3P PDST = 3-6P EDST = 19-22 UTC 29 Tues 5-6P PDST = 8-9P EDST = 0- 1 UTC Wed 30 USA-PacificDaylightSavingsTime is -7 hours UTC, due to daylight savings currently. Times listed above should be double checked by you for accuracy. ===== 3) LOCAL MEETING AT U. C. BERKELEY - LOCATION TO BE DETERMINED http://sites.google.com/site/berkeleytip/directions RSVP please. See below. It greatly helps my planning. But, _do_ come if you forgot to RSVP. THE JUNE 5 & 20 ON CAMPUS MEETING LOCATIONS ARE CURRENTLY TO BE DETERMINED. HOPEFULLY KNOW ON FRIDAY JUNE 4. PLEASE VIEW THE BTIP WEBSITE & MAILING LIST FOR THE FINALIZED LOCATION. THANK YOU. ALWAYS BE SURE TO CHECK THE BTIP WEBSITE _&_ MAILING LIST FOR THE LATEST LAST MINUTE DETAILS & CHANGES, BEFORE COMING TO THE MEETING! :) http://sites.google.com/site/berkeleytip http://groups.google.com/group/BerkTIPGlobal DO BRING A VOIP HEADSET, available for $10-30 at most electronics retail stores, & a laptop computer, so you are able to communicate with the global BTIP community via VOIP. It is highly recommended that you have a voip headset, & not rely on a laptop's built in microphone & speakers, because the headphones keep the noise level down. Bringing a headset is not required, but is a great part of the being able to communicate with the global community. :) Clothing: Typically 55-80 degrees F. Weather: http://www.wunderground.com/auto/sfgate/CA/Berkeley.html Other location local meeting possibilities: http://sites.google.com/site/berkeleytip/local-meetings Create a local meeting in your town. ===== 4) HOT TOPICS Oracle owns Sun - Free SW implications? OpenOffice? OpenOffice - Ready for college Fall 2010? MakerBot, RepRap - Personal Making - Like where PCs were in 1975? Android phones - Besting iPhone? worthwhile? How knowable is the hw? - Can BSD be run on Android phones? iPad, iPhone & iPod- rooting & running GNU(Linux) & BSD ===== 5) PLEASE RSVP PROBABILISTICALLY, THANKS :) If you think there is a >70% chance ("likely") you'll come to the in person meeting in Berkeley, please RSVP to me. Thanks. It helps my planning. Please _do_ come even if you haven't RSVP'd, it's not required. Better yet, join the BerkeleyTIP-Global mailing list, send the RSVP there, & tell us what things you're interested in, or what videos you'll try to watch - so we can know what videos are popular, & we might watch them too. :) http://groups.google.com/group/BerkTIPGlobal ===== 6) INSTALLFEST Get help installing & using Free Software, Hardware & Culture. Laptops only, typically. There isn't easy access for physically bringing desktop boxes here. RSVP _HIGHLY RECOMMENDED_ if you want installfest help. Please RSVP to me, Giovanni, at the from address for this announcement, or better, join & send email to the BTIP-Global mailing list telling us what you'd like help with. This way we can be better prepared to help you, & you might get valuable advice from the mailing list members. If you are new to using free software, an excellent system would be the KUbuntu GNU(Linux) software. It is very comprehensive, fairly easy to use (similar to Windows or Mac), & suitable for personal, home, university, or business use. We are also glad to try to help people with software who join via VOIP. Please email the mailing list with requests that you want help with, so we can try to be prepared better to help you. Installfest volunteers/helpers always welcome, in person, or via VOIP. :) ===== SPECIAL - MUSIC IN BERKELEY JUNE 5 SATURDAY ===== Also June 5 Sat in Berkeley, after BTIP: ===== FREE 7th Annual BERKELEY WORLD MUSIC FESTIVAL ===== Telegraph Avenue 12 Noon - 9 pm "Berkeley World Music Fest... has some of our best world musicians who call East Bay home...a wonderful event, and intimate, mix of outdoor performances in cafes and shops(the best music People's Park gets each year along Telegraph Ave. (Larry Kelp, KPFA music host)" Continuous music outdoors & in cafes http://www.berkeleyworldmusic.com/entry.asp?PageID=118 ===== 7) ARRIVING FIRST AT THE MEETING: MAKE A "BerkeleyTIP" SIGN If you get to the meeting & don't see a "BerkeleyTIP" sign up yet, please: 1) Make a BTIP sign on an 8x11 paper & put it at your table, 2) Email the mailing list, or join on IRC, & let us know you are there. Ask someone if you could use their computer for a minute to look something up, or send an email. People are usually very friendly & willing to help. We can also email you a temporary guest AirBears account login. We will have wifi guest accounts available for BTIP attendees. Be sure you have wifi capable equipment. Be Prepared: Bring a multi-outlet extension power cord. ===== 8) IRC: #berkeleytip on irc.freenode.net For help with anything, especially how to get VOIP working, & text communication. ===== 9) VOIP FOR GLOBAL MEETING Speak & listen to everyone globally using VOIP. Get a headset! See some VOIP instructions here: http://sites.google.com/site/berkeleytip/voice-voip-conferencing ===== 10) VOLUNTEERING, TO DOs Enjoy doing or learning something(s)? Help out BTIP in that area. Website development, mailing list management, video locating, VOIP server (FreeSwitch, Asterisk) or client (Ekiga, SFLPhone,...), creating a local meeting. Join the mailing list & let us know. Your offers of free help are always welcome here. :) ===== 11) MAILING LISTS: BerkeleyTIP-Global, LocalBerkeley, Announce Everyone should join the BerkeleyTIP-Global list: http://groups.google.com/group/BerkTIPGlobal Say "hi", tell us your interests, & what videos you'll like to watch. Info on all lists here: http://sites.google.com/site/berkeleytip/mailing-lists ===== 12) ANYTHING I FORGOT TO MENTION? Please join & email the BerkeleyTIP-Global mailing list. ===== 13) FOR FORWARDING You are invited to forward this message anywhere it would be appreciated. Better yet, use it to create a local meeting. Invite & get together with your friends locally, & join in with us all globally. :) Looking forward to meeting with you in person, or online. :) Giovanni From steveayre at gmail.com Fri Jun 4 05:13:02 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 4 Jun 2010 13:13:02 +0100 Subject: [Freeswitch-users] outbound socket problem / fs_ivrd general protection, segfault In-Reply-To: <4C08C3E2.3020703@xpirio.com> References: <4C08C3E2.3020703@xpirio.com> Message-ID: Hi Christian, If it's segfaulting, then there is a bug somewhere. Firstly, make sure you're on the very latest version from Git. See if you can reproduce the error on that version. If you're on an old version then there's a possibility the bug has already been fixed. If you still have segfault problems collect a core dump file (which may require running 'ulimit -c unlimited' before starting fs_ivrd) for the process and report it with the backtrace from gdb on Jira ( http://jira.freeswitch.org/). Load the core file into gdb with the command 'gdb fs_ivrd core', and post the output for the commands 'bt full' and 'thread apply all bt full'. See http://wiki.freeswitch.org/wiki/Reporting_Bugs -Steve 2010/6/4 Christian L?schenkohl > hello list > > recently i have problems with outbound socket scripts (written in php) > > dialplan looks like this > > > data="ivr_path=/opt/freeswitch/scripts/script1.php"/> > > ... do something ... > data="ivr_path=/opt/freeswitch/scripts/script2.php"/> > > > > > script1.php is executed script2.php fails (see syslog msg) > > syslog says > fs_ivrd[11971] general protection ip:7f2f7b94bbf5 sp:7fffffffa148 error:0 > in libpthread-2.7.so[7f2f7b942000+16000] or > fs_ivrd[12037]: segfault at a006c6c7576 ip 7f2f7b94bbf5 sp 7fffffffa148 > error 4 in libpthread-2.7.so[7f2f7b942000+16000] > > uname -a > Linux vts02.vie.xpirio.net 2.6.26-2-amd64 #1 SMP Tue Mar 9 22:29:32 UTC > 2010 x86_64 GNU/Linux > > version > FreeSWITCH Version 1.0.6 (svn-exported) > > does anybody have similar problems? > > br > > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/6fdc2c02/attachment.html From infos at madovsky.org Fri Jun 4 06:59:17 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 4 Jun 2010 09:59:17 -0400 Subject: [Freeswitch-users] check status of a number References: <9833524A0669475FA86895E58A23CF16@MOBILEE1705><85E0E8E1683646568D9E1929D7CD1108@MOBILEE1705> Message-ID: <651C0E8350F9446B8509F356C784D7BF@MOBILEE1705> ok I understand Thanks Franck ----- Original Message ----- From: Steven Ayre To: freeswitch-users at lists.freeswitch.org Sent: Friday, June 04, 2010 3:38 AM Subject: Re: [Freeswitch-users] check status of a number It means the originate command succeeded to create a call, which that UUID identifies. It doesn't tell you anything about the result of that call though - you'll need to watch for events for that UUID, or check its XML CDR to find out the callflow (to see if it matched an extension) or the hangup reason. -Steve On 4 June 2010 04:51, Madovsky wrote: Ok I succeed doing like this : /usr/local/freeswitch/bin/fs_cli -x "expand originate sofia/gateway/\${distributor(pstn_international_2)}/003434REMOTEEXT loopback/999LOCALEXT" and it gives a session: +OK d0c598b5-9291-47eb-a14e-7845763425f4 does +OK means the number always exists ? is there only one ring, hangup and clean the originate call ? Thanks F ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 03, 2010 2:22 PM Subject: Re: [Freeswitch-users] check status of a number Use the loopback endpoint with originate to do this. /b On Jun 3, 2010, at 1:16 PM, Madovsky wrote: Hi, I'd like to use fs_cli to check if the callee number exists and get the status result. Is anyone already did that ? Thanks Franck -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/7526b8f1/attachment-0001.html From brian at freeswitch.org Fri Jun 4 07:28:35 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Jun 2010 09:28:35 -0500 Subject: [Freeswitch-users] Help debugging INVITE with provider In-Reply-To: <404ccbd42e100f414bab82b5bd65461e@thom.fr.eu.org> References: <07314285-F095-4946-B041-D50CD875326D@gmail.com> <404ccbd42e100f414bab82b5bd65461e@thom.fr.eu.org> Message-ID: <8C40FBE2-46D2-4881-83D1-502713FEFABE@freeswitch.org> Well looking at the invite from your provider: v=0 o=FrancoisLegal 1275590034 1275590034 IN IP4 80.239.235.114 s=SIP Call c=IN IP4 80.239.235.114 t=0 0 m=audio 11198 RTP/AVP 2 3 9 8 0 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 11340 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 <-- THIS LINE a=rtpmap:99 <-- AND THIS LINE a=ptime:20 RFC4566 section 6: RTP profiles that specify the use of dynamic payload types MUST define the set of valid encoding names and/or a means to register encoding names if that profile is to be used with SDP. We are left guessing what 99 and 121 are which is invalid. /b On Jun 4, 2010, at 2:57 AM, Fran?ois Legal wrote: > This is what I did (except for the last trace) : MyProviderIp is the IP FS tries to contact to reach SomeAccount at provider.com, and MyFSIpHere is the IP FS external profile is bound to. > > Maybe this lack some information : the "dialed number" is originate sofia/external/SomeAccount at provider.com > > which should at the end come back to my FS external profile through the gateway 12voip.com included in the external profile. > > > Fran?ois > From christian.loeschenkohl at xpirio.com Fri Jun 4 07:39:34 2010 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Fri, 04 Jun 2010 16:39:34 +0200 Subject: [Freeswitch-users] outbound socket problem / fs_ivrd general protection, segfault In-Reply-To: References: <4C08C3E2.3020703@xpirio.com> Message-ID: <4C091026.5040209@xpirio.com> hi thank you - i use a very up-to-date version - see version output 1.0.6 the ivrd hasn't changed from 1.0.6 to the current trunk - i don't know how to debug with gdb here freeswitch dumps it's core as a core file but fs_ivrd doesn't could you provide a command that does the job? strace also syslog says the kernel logs this Jun 4 14:23:43 vts02 kernel: [2321314.877379] fs_ivrd[18070] general protection ip:7f2f7b94bbf5 sp:7fffffffa148 error:0 in libpthread-2.7.so[7f2f7b942000+16000] or Jun 4 14:47:53 vts02 kernel: [2323898.324458] fs_ivrd[19023]: segfault at a006c6c85 ip 7f0429f43bf5 sp 7fffffffab68 error 4 in libpthread-2.7.so[7f0429f3a000+16000] fs_ivrd doesn't break and it does not need a restart br On 2010-06-04 14:13, Steven Ayre wrote: > Hi Christian, > > If it's segfaulting, then there is a bug somewhere. > > Firstly, make sure you're on the very latest version from Git. See if > you can reproduce the error on that version. If you're on an old version > then there's a possibility the bug has already been fixed. > > If you still have segfault problems collect a core dump file (which may > require running 'ulimit -c unlimited' before starting fs_ivrd) for the > process and report it with the backtrace from gdb on Jira > (http://jira.freeswitch.org/). Load the core file into gdb with the > command 'gdb fs_ivrd core', and post the output for the commands 'bt > full' and 'thread apply all bt full'. > > See http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -Steve > > > > 2010/6/4 Christian L?schenkohl > > > hello list > > recently i have problems with outbound socket scripts (written in php) > > dialplan looks like this > > break="on-true"> > data="ivr_path=/opt/freeswitch/scripts/script1.php"/> > > ... do something ... > data="ivr_path=/opt/freeswitch/scripts/script2.php"/> > > > > > script1.php is executed script2.php fails (see syslog msg) > > syslog says > fs_ivrd[11971] general protection ip:7f2f7b94bbf5 sp:7fffffffa148 > error:0 in libpthread-2.7.so > [7f2f7b942000+16000] or > fs_ivrd[12037]: segfault at a006c6c7576 ip 7f2f7b94bbf5 sp > 7fffffffa148 error 4 in libpthread-2.7.so > [7f2f7b942000+16000] > > uname -a > Linux vts02.vie.xpirio.net > 2.6.26-2-amd64 #1 SMP Tue Mar 9 22:29:32 UTC 2010 x86_64 GNU/Linux > > version > FreeSWITCH Version 1.0.6 (svn-exported) > > does anybody have similar problems? > > br > > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From brian at freeswitch.org Fri Jun 4 07:44:06 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Jun 2010 09:44:06 -0500 Subject: [Freeswitch-users] outbound socket problem / fs_ivrd general protection, segfault In-Reply-To: <4C091026.5040209@xpirio.com> References: <4C08C3E2.3020703@xpirio.com> <4C091026.5040209@xpirio.com> Message-ID: <7F6151C9-4819-471E-B962-552DDB324253@freeswitch.org> Are you using GIT to pull the src? /b On Jun 4, 2010, at 9:39 AM, Christian L?schenkohl wrote: > hi > > thank you > > - i use a very up-to-date version - see version output 1.0.6 > the ivrd hasn't changed from 1.0.6 to the current trunk > > - i don't know how to debug with gdb here > freeswitch dumps it's core as a core file but fs_ivrd doesn't > could you provide a command that does the job? > strace > > also syslog says the kernel logs this > Jun 4 14:23:43 vts02 kernel: [2321314.877379] fs_ivrd[18070] general protection ip:7f2f7b94bbf5 sp:7fffffffa148 error:0 in libpthread-2.7.so[7f2f7b942000+16000] > or > Jun 4 14:47:53 vts02 kernel: [2323898.324458] fs_ivrd[19023]: segfault at a006c6c85 ip 7f0429f43bf5 sp 7fffffffab68 error 4 in libpthread-2.7.so[7f0429f3a000+16000] > > fs_ivrd doesn't break and it does not need a restart > > br From yeukfung at gmail.com Thu Jun 3 22:02:32 2010 From: yeukfung at gmail.com (Russell Kwok) Date: Fri, 4 Jun 2010 13:02:32 +0800 Subject: [Freeswitch-users] Unable to detect the digits in demo IVR when connecting via DID inbound Message-ID: Hello all, Sorry to bother all, I had been trying so hard to create a DID-inbound that will route to demo IVR(5000 XML default) and allow user to press and select the menu via pressing digits. It works perfect when I dial 5000 from any of my local extension. However, when I try to call the DID, and got connected to the voice mail successfully, but the digit is unable to be detected. Wondering if my configuration got something wrong? what i had created in the public\01_mydivert.xml (i put the config inside the public context and setting mydivert to connect to my pbxserver at port 5080) file is as follow: data="transfer_ringback=$${uk_ring}" /> and when i call the DID number hosted by mydivert.com, the log file shows when i press 1, it just do nothing and keep playing to "please press 2 or something" 2010-06-04 12:23:59.414516 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[voicemail/vm-press.wav] (en:en) 2010-06-04 12:23:59.414516 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms 2010-06-04 12:23:59.814618 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:23:59.933629 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[digits/1.wav] (en:en) 2010-06-04 12:23:59.933629 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms 2010-06-04 12:24:00.394005 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:24:00.514102 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[ivr/ivr-to_do_a_freeswitch_echo_test.wav] (en:en) 2010-06-04 12:24:00.514102 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms > 2010-06-04 12:24:01.596587 [DEBUG] switch_rtp.c:2841 RTP RECV DTMF 1:1280 2010-06-04 12:24:01.599459 [DEBUG] zap_io.c:2061 4:1 GENERATE DTMF [1] 2010-06-04 12:24:01.639329 [DEBUG] mod_openzap.c:721 queue DTMF [1] > 2010-06-04 12:24:01.654270 [DEBUG] switch_rtp.c:1928 Send start packet for > [1] ts=172800 dur=160/160/2000 seq=63824 2010-06-04 12:24:01.674124 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=320/320/2000 seq=63825 2010-06-04 12:24:01.694115 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=480/480/2000 seq=63826 2010-06-04 12:24:01.714098 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=640/640/2000 seq=63827 2010-06-04 12:24:01.733685 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=800/800/2000 seq=63828 2010-06-04 12:24:01.754129 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=960/960/2000 seq=63829 2010-06-04 12:24:01.774264 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1120/1120/2000 seq=63830 2010-06-04 12:24:01.795396 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1280/1280/2000 seq=63831 2010-06-04 12:24:01.814112 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1440/1440/2000 seq=63832 2010-06-04 12:24:01.834315 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1600/1600/2000 seq=63833 2010-06-04 12:24:01.853792 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1760/1760/2000 seq=63834 2010-06-04 12:24:01.874010 [DEBUG] switch_rtp.c:1864 Send middle packet for > [1] ts=172800 dur=1920/1920/2000 seq=63835 2010-06-04 12:24:01.894444 [DEBUG] switch_rtp.c:1864 Send end packet for [1] > ts=172800 dur=2080/2080/2000 seq=63836 2010-06-04 12:24:01.894444 [DEBUG] switch_rtp.c:1864 Send end packet for [1] > ts=172800 dur=2080/2080/2000 seq=63837 2010-06-04 12:24:01.894444 [DEBUG] switch_rtp.c:1864 Send end packet for [1] > ts=172800 dur=2080/2080/2000 seq=63838 2010-06-04 12:24:02.333695 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:24:02.434126 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[ivr/ivr-please.wav] (en:en) 2010-06-04 12:24:02.434126 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms 2010-06-04 12:24:02.854127 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:24:02.974217 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[voicemail/vm-press.wav] (en:en) 2010-06-04 12:24:02.974217 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms 2010-06-04 12:24:03.374396 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:24:03.494367 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[digits/2.wav] (en:en) 2010-06-04 12:24:03.494367 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms and the app info for this DID inbound is as follow: 2010-06-04 12:23:41.001560 [INFO] mod_dptools.c:973 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/24172200 at 78.46.23.202] Unique-ID: [f456ff9c-6f90-11df-84c8-3b7d920feee8] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [24172200] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [24172200] Caller-Caller-ID-Number: [24172200] Caller-Network-Addr: [78.46.23.202] Caller-ANI: [24172200] Caller-Destination-Number: [85236931541] Caller-Unique-ID: [f456ff9c-6f90-11df-84c8-3b7d920feee8] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/external/24172200 at 78.46.23.202] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1275625420968944] Caller-Channel-Created-Time: [1275625420968944] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [false] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_direction: [inbound] variable_uuid: [f456ff9c-6f90-11df-84c8-3b7d920feee8] variable_sip_local_network_addr: [192.168.8.232] variable_sip_network_ip: [78.46.23.202] variable_sip_network_port: [5060] variable_sip_received_ip: [78.46.23.202] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [24172200] variable_sip_from_uri: [24172200 at 78.46.23.202] variable_sip_from_host: [78.46.23.202] variable_sip_from_user_stripped: [24172200] variable_sip_from_tag: [as750696c8] variable_sofia_profile_name: [external] variable_sip_Remote-Party-ID: ["24172200" > >;privacy=off;screen=no] variable_sip_cid_type: [rpid] variable_sip_full_via: [SIP/2.0/UDP 78.46.23.202:5060 > ;branch=z9hG4bK32bcbf22] variable_sip_from_display: [24172200] variable_sip_full_from: ["24172200" > >;tag=as750696c8] variable_sip_full_to: [] variable_sip_req_user: [85236931541] variable_sip_req_port: [5080] variable_sip_req_uri: [85236931541 at jessnruss.dyndns.org:5080] variable_sip_req_host: [jessnruss.dyndns.org] variable_sip_to_user: [85236931541] variable_sip_to_port: [5080] variable_sip_to_uri: [85236931541 at jessnruss.dyndns.org:5080] variable_sip_to_host: [jessnruss.dyndns.org] variable_sip_contact_user: [24172200] variable_sip_contact_uri: [24172200 at 78.46.23.202] variable_sip_contact_host: [78.46.23.202] variable_channel_name: [sofia/external/24172200 at 78.46.23.202] variable_sip_call_id: [6b53813d4bb32def4d2931960d17e7ee at 78.46.23.202] variable_sip_user_agent: [Asterisk] variable_sip_via_host: [78.46.23.202] variable_sip_via_port: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=root 1913419625 1913419625 IN IP4 78.46.23.202 s=Asterisk PBX 1.6.0.26 c=IN IP4 78.46.23.202 t=0 0 m=audio 19066 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ] variable_remote_media_ip: [78.46.23.202] variable_remote_media_port: [19066] variable_sip_use_codec_name: [PCMU] variable_sip_use_codec_rate: [8000] variable_sip_use_codec_ptime: [20] variable_read_codec: [PCMU] variable_read_rate: [8000] variable_write_codec: [PCMU] variable_write_rate: [8000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_current_application: [info] However, if i call the demo IVR 5000 internally, the log shows really good. 2010-06-04 12:30:22.394100 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[voicemail/vm-press.wav] (en:en) 2010-06-04 12:30:22.394100 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms 2010-06-04 12:30:22.794457 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:30:22.914491 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[digits/1.wav] (en:en) 2010-06-04 12:30:22.914491 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms 2010-06-04 12:30:23.373688 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:30:23.474363 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[ivr/ivr-to_do_a_freeswitch_echo_test.wav] (en:en) 2010-06-04 12:30:23.474363 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms > 2010-06-04 12:30:25.215382 [DEBUG] switch_rtp.c:2841 RTP RECV DTMF 1:960 2010-06-04 12:30:25.215382 [DEBUG] switch_ivr_play_say.c:1444 done playing > file 2010-06-04 12:30:25.334284 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4 > digits t/o 2000 2010-06-04 12:30:27.354514 [DEBUG] switch_ivr_menu.c:376 digits '1' 2010-06-04 12:30:27.354514 [DEBUG] switch_ivr_menu.c:470 action regex [1] > [/^(10[01][0-9])$/] [0] > 2010-06-04 12:30:27.354514 [DEBUG] switch_ivr_menu.c:488 IVR action on menu > 'demo_ivr' matched '1' param 'bridge sofia/ > 192.168.8.232/888 at conference.freeswitch.org' > 2010-06-04 12:30:27.354514 [DEBUG] switch_ivr_menu.c:492 > switch_ivr_menu_execute todo=[2] EXECUTE sofia/internal/1001 at jessnruss.dyndns.org bridge(sofia/ > 192.168.8.232/888 at conference.freeswitch.org) from the two logs above, what i can see the difference is the line: in [FAILURE log] 2010-06-04 12:24:01.596587 [DEBUG] switch_rtp.c:2841 RTP RECV DTMF 1:1280 in [SUCCESS log] 2010-06-04 12:30:25.215382 [DEBUG] switch_rtp.c:2841 RTP RECV DTMF 1:960 i am wondering if is there anything to do with this line? any help is really appreciated. Cheers, Russell Kwok -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/e2760113/attachment-0001.html From thangappan143 at gmail.com Fri Jun 4 02:01:30 2010 From: thangappan143 at gmail.com (Thangappan.M) Date: Fri, 4 Jun 2010 14:31:30 +0530 Subject: [Freeswitch-users] Need to stop more than one voice file using break application In-Reply-To: References: Message-ID: Dear all, I am in the process of developing IVR using FreeSWITCH. For that I am being used outbound ESL in async mode. In my design, usually for one menu it might be more than one voice files. So using playback_delimiter, play back all the voice file in single instance using playback application. I've tried to stop the play back using "break all" API. But it only break the only one voice file not the whole application(playback). Consider that I am playback four voice files. When the first voice file is getting playback, using " break all" t stop the playback . It only stopped the second voice and continued to playback the third voice file and followed by fourth one. My need is to break the playback application which may have any number of voice files in async mode. Thanks in advance. -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/2e96195d/attachment.html From hitormiz13 at yahoo.com Fri Jun 4 02:46:20 2010 From: hitormiz13 at yahoo.com (ro ra) Date: Fri, 4 Jun 2010 17:46:20 +0800 (SGT) Subject: [Freeswitch-users] Registration Update of Contact List Message-ID: <69272.33850.qm@web76807.mail.sg1.yahoo.com> Hi, ??? what do i need to change on my configuration files to make? this scenario work: ??? user1?? --------- REGISTER????? ------------> Freeswitch ??? user1? <-------- 401 UnAuthorized -------?? Freeswitch ??? user1 ? --------- REGISTER w/ Auth -----> Freeswitch ??? user1?? <------- 200 OK? ------------------------?? Freeswitch ??? ----------------------------5 secs delay---------------------------------- ??? user1 ----------- REGISTER w/ Auth(Update) ----->? Freeswitch ??? user1 <-------- 200 OK?? -----------------------???? Freeswitch ?? or is this possible with freeswitch...? ??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/239b0cf9/attachment.html From devel at thom.fr.eu.org Fri Jun 4 07:56:58 2010 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Fri, 04 Jun 2010 16:56:58 +0200 Subject: [Freeswitch-users] Help debugging INVITE with provider In-Reply-To: <8C40FBE2-46D2-4881-83D1-502713FEFABE@freeswitch.org> References: <07314285-F095-4946-B041-D50CD875326D@gmail.com> <404ccbd42e100f414bab82b5bd65461e@thom.fr.eu.org> <8C40FBE2-46D2-4881-83D1-502713FEFABE@freeswitch.org> Message-ID: <066d6f86556d985bcb0d7ad47d2c763c@thom.fr.eu.org> Thanks for the help. I could fix it by setting an external profile specific outbound_codec_list not including video codecs (as I don't expect video to come from my providers) and that did the trick. Fran?ois On Fri, 4 Jun 2010 09:28:35 -0500, Brian West wrote: > Well looking at the invite from your provider: > > v=0 > o=FrancoisLegal 1275590034 1275590034 IN IP4 80.239.235.114 > s=SIP Call > c=IN IP4 80.239.235.114 > t=0 0 > m=audio 11198 RTP/AVP 2 3 9 8 0 101 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > m=video 11340 RTP/AVP 31 34 115 121 99 > a=rtpmap:31 H261/90000 > a=rtpmap:34 H263/90000 > a=rtpmap:115 H263-1998/90000 > a=rtpmap:121 <-- THIS LINE > a=rtpmap:99 <-- AND THIS LINE > a=ptime:20 > > RFC4566 section 6: > > RTP profiles that specify the use of dynamic payload types MUST define the > set of valid encoding names and/or a means to register encoding names if > that profile is to be used with SDP. > > We are left guessing what 99 and 121 are which is invalid. > > /b > > > On Jun 4, 2010, at 2:57 AM, Fran?ois Legal wrote: > >> This is what I did (except for the last trace) : MyProviderIp is the IP >> FS tries to contact to reach SomeAccount at provider.com, and MyFSIpHere is >> the IP FS external profile is bound to. >> >> Maybe this lack some information : the "dialed number" is originate >> sofia/external/SomeAccount at provider.com >> >> which should at the end come back to my FS external profile through the >> gateway 12voip.com included in the external profile. >> >> >> Fran?ois >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brian at freeswitch.org Fri Jun 4 07:59:32 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Jun 2010 09:59:32 -0500 Subject: [Freeswitch-users] Registration Update of Contact List In-Reply-To: <69272.33850.qm@web76807.mail.sg1.yahoo.com> References: <69272.33850.qm@web76807.mail.sg1.yahoo.com> Message-ID: <76515BA8-C8D1-4C62-9587-A438E1714A75@freeswitch.org> It should already do this if your user1 device supports the proper nonce count. /b On Jun 4, 2010, at 4:46 AM, ro ra wrote: > Hi, > > what do i need to change on my configuration files to make this scenario work: > > user1 --------- REGISTER ------------> Freeswitch > user1 <-------- 401 UnAuthorized ------- Freeswitch > user1 --------- REGISTER w/ Auth -----> Freeswitch > user1 <------- 200 OK ------------------------ Freeswitch > ----------------------------5 secs delay---------------------------------- > user1 ----------- REGISTER w/ Auth(Update) -----> Freeswitch > user1 <-------- 200 OK ----------------------- Freeswitch > > or is this possible with freeswitch...? > > thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/6d252475/attachment.html From christian.loeschenkohl at xpirio.com Fri Jun 4 08:04:57 2010 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Fri, 04 Jun 2010 17:04:57 +0200 Subject: [Freeswitch-users] outbound socket problem / fs_ivrd general protection, segfault In-Reply-To: <7F6151C9-4819-471E-B962-552DDB324253@freeswitch.org> References: <4C08C3E2.3020703@xpirio.com> <4C091026.5040209@xpirio.com> <7F6151C9-4819-471E-B962-552DDB324253@freeswitch.org> Message-ID: <4C091619.9020200@xpirio.com> hello the running version is http://files.freeswitch.org/freeswitch-1.0.6.tar.bz2 i compared the ivrd source with the version i did get with git clone git://git.freeswitch.org/freeswitch.git br On 2010-06-04 16:44, Brian West wrote: > Are you using GIT to pull the src? > > /b > > On Jun 4, 2010, at 9:39 AM, Christian L?schenkohl wrote: > >> hi >> >> thank you >> >> - i use a very up-to-date version - see version output 1.0.6 >> the ivrd hasn't changed from 1.0.6 to the current trunk >> >> - i don't know how to debug with gdb here >> freeswitch dumps it's core as a core file but fs_ivrd doesn't >> could you provide a command that does the job? >> strace >> >> also syslog says the kernel logs this >> Jun 4 14:23:43 vts02 kernel: [2321314.877379] fs_ivrd[18070] general protection ip:7f2f7b94bbf5 sp:7fffffffa148 error:0 in libpthread-2.7.so[7f2f7b942000+16000] >> or >> Jun 4 14:47:53 vts02 kernel: [2323898.324458] fs_ivrd[19023]: segfault at a006c6c85 ip 7f0429f43bf5 sp 7fffffffab68 error 4 in libpthread-2.7.so[7f0429f3a000+16000] >> >> fs_ivrd doesn't break and it does not need a restart >> >> br > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From brian at freeswitch.org Fri Jun 4 08:13:44 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Jun 2010 10:13:44 -0500 Subject: [Freeswitch-users] Help debugging INVITE with provider In-Reply-To: <066d6f86556d985bcb0d7ad47d2c763c@thom.fr.eu.org> References: <07314285-F095-4946-B041-D50CD875326D@gmail.com> <404ccbd42e100f414bab82b5bd65461e@thom.fr.eu.org> <8C40FBE2-46D2-4881-83D1-502713FEFABE@freeswitch.org> <066d6f86556d985bcb0d7ad47d2c763c@thom.fr.eu.org> Message-ID: You can't fix this... the invite is coming from your provider with those in it.. and its invalid so you have no way to correct this unless you install a proxy sever and rewrite the SDP before you send it to freeswitch... your provder's VERY NICE sip proxy isn't so very NICE. /b On Jun 4, 2010, at 9:56 AM, Fran?ois Legal wrote: > Thanks for the help. > > I could fix it by setting an external profile specific outbound_codec_list > not including video codecs (as I don't expect video to come from my > providers) and that did the trick. > > Fran?ois From brian at freeswitch.org Fri Jun 4 08:14:12 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Jun 2010 10:14:12 -0500 Subject: [Freeswitch-users] outbound socket problem / fs_ivrd general protection, segfault In-Reply-To: <4C091619.9020200@xpirio.com> References: <4C08C3E2.3020703@xpirio.com> <4C091026.5040209@xpirio.com> <7F6151C9-4819-471E-B962-552DDB324253@freeswitch.org> <4C091619.9020200@xpirio.com> Message-ID: <19DAE4B7-238F-4C77-A8A6-4B46B12A2E00@freeswitch.org> Can you create a small test case that someone like myself could have to demonstrate this issue? /b On Jun 4, 2010, at 10:04 AM, Christian L?schenkohl wrote: > hello > > the running version is http://files.freeswitch.org/freeswitch-1.0.6.tar.bz2 > > i compared the ivrd source with the version i did get with > git clone git://git.freeswitch.org/freeswitch.git > > br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/9206bcde/attachment.html From anthony.minessale at gmail.com Fri Jun 4 08:18:55 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Jun 2010 10:18:55 -0500 Subject: [Freeswitch-users] outbound socket problem / fs_ivrd general protection, segfault In-Reply-To: <4C091619.9020200@xpirio.com> References: <4C08C3E2.3020703@xpirio.com> <4C091026.5040209@xpirio.com> <7F6151C9-4819-471E-B962-552DDB324253@freeswitch.org> <4C091619.9020200@xpirio.com> Message-ID: it's not just about ivrd its about the whole esl library and everything that uses it. Every time you update FS you must also update all of the ESL and ESL modules 2010/6/4 Christian L?schenkohl > hello > > the running version is > http://files.freeswitch.org/freeswitch-1.0.6.tar.bz2 > > i compared the ivrd source with the version i did get with > git clone git://git.freeswitch.org/freeswitch.git > > br > > On 2010-06-04 16:44, Brian West wrote: > > > Are you using GIT to pull the src? > > > > /b > > > > On Jun 4, 2010, at 9:39 AM, Christian L?schenkohl wrote: > > > >> hi > >> > >> thank you > >> > >> - i use a very up-to-date version - see version output 1.0.6 > >> the ivrd hasn't changed from 1.0.6 to the current trunk > >> > >> - i don't know how to debug with gdb here > >> freeswitch dumps it's core as a core file but fs_ivrd doesn't > >> could you provide a command that does the job? > >> strace > >> > >> also syslog says the kernel logs this > >> Jun 4 14:23:43 vts02 kernel: [2321314.877379] fs_ivrd[18070] general > protection ip:7f2f7b94bbf5 sp:7fffffffa148 error:0 in libpthread-2.7.so > [7f2f7b942000+16000] > >> or > >> Jun 4 14:47:53 vts02 kernel: [2323898.324458] fs_ivrd[19023]: segfault > at a006c6c85 ip 7f0429f43bf5 sp 7fffffffab68 error 4 in libpthread-2.7.so > [7f0429f3a000+16000] > >> > >> fs_ivrd doesn't break and it does not need a restart > >> > >> br > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100604/dbb5c5ec/attachment-0001.html From jan.berger at video24.no Fri Jun 4 08:22:17 2010 From: jan.berger at video24.no (Jan Berger) Date: Fri, 4 Jun 2010 17:22:17 +0200 Subject: [Freeswitch-users] Need to stop more than one voice file using break application In-Reply-To: References: Message-ID: <0AB5E608E3464DCFB86C64279CF634DE@dell9400> Yupp I have this on vxml as well - also checked code earlier the barge-in in FS is local for what you do here and now - next PlayFile reset it. Use playphrase - that allows you to play several files in one go. Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Thangappan.M Sent: 4. juni 2010 11:02 To: freeswitch-users Subject: [Freeswitch-users] Need to stop more than one voice file using break application Dear all, I am in the process of developing IVR using FreeSWITCH. For that I am being used outbound ESL in async mode. In my design, usually for one menu it might be more than one voice files. So using playback_delimiter, play back all the voice file in single instance using playback application. I've tried to stop the play back using "break all" API. But it only break the only one voice file not the whole application(playback). Consider that I am playback four voice files. When the first voice file is getting playback, using " break all" t stop the playback . It only stopped the second voice and continued to playback the third voice file and followed by fourth one. My need is to break the playback application which may have any number of voice files in async mode. Thanks in advance. -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/f8b06e64/attachment.html From anthony.minessale at gmail.com Fri Jun 4 08:33:30 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Jun 2010 10:33:30 -0500 Subject: [Freeswitch-users] Need to stop more than one voice file using break application In-Reply-To: References: Message-ID: I added a patch [eba05c3 ] so it will work now On Fri, Jun 4, 2010 at 4:01 AM, Thangappan.M wrote: > > Dear all, > > I am in the process of developing IVR using FreeSWITCH. For that I am > being used outbound ESL in async mode. > > In my design, usually for one menu it might be more than one voice > files. So using playback_delimiter, play back all the voice file in single > instance using playback application. > > I've tried to stop the play back using "break all" API. But it > only break the only one voice file not the whole application(playback). > > Consider that I am playback four voice files. When the first voice file > is getting playback, using " break all" t stop the playback . It > only stopped the second voice and continued to playback the third voice file > and followed by fourth one. > > My need is to break the playback application which may have any number > of voice files in async mode. > > Thanks in advance. > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100604/2ee03434/attachment.html From pjintheusa at gmail.com Fri Jun 4 08:53:17 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 4 Jun 2010 11:53:17 -0400 Subject: [Freeswitch-users] Freeswicth nms In-Reply-To: References: <1111639352-1275526834-cardhu_decombobulator_blackberry.rim.net-2040530345-@bda2058.bisx.prodap.on.blackberry> Message-ID: Are there any FreeSWITCH specific plugins available for Nagios? On Thu, Jun 3, 2010 at 10:43 PM, Code Ghar wrote: > I would also recommend Nagios. Although we are not doing anything funky, it > has saved our skin multiple times. It is fairly easy to use with lots of > plugins. I haven't used it with FS, though. Whatever you use please try to > share your config, etc. > > > On Wed, Jun 2, 2010 at 8:00 PM, wrote: > >> Budi, >> Try Nagios for network monitoring >> >> Regards >> >> Suvir >> ------Original Message------ >> From: Budi wibowo >> Sender: freeswitch-users-bounces at lists.freeswitch.org >> To: freeswitch-users at lists.freeswitch.org >> ReplyTo: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Freeswicth nms >> Sent: Jun 3, 2010 06:26 >> >> Hi all >> I'm looking for nms that can do minimum alarm,fault, performance and >> statistic. Alarm could be sent via sms or email. >> I need the such solution due I have project to involve more than 10 fs >> box. >> Both free and commercial solution are ok >> >> Regards >> Budi >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> Sent via BlackBerry? from AIS >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100604/740397dd/attachment.html From infos at madovsky.org Fri Jun 4 09:01:53 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 4 Jun 2010 12:01:53 -0400 Subject: [Freeswitch-users] check status of a number References: <9833524A0669475FA86895E58A23CF16@MOBILEE1705><85E0E8E1683646568D9E1929D7CD1108@MOBILEE1705> Message-ID: <9C950B09DD16465DB3C19D79D6CFB71E@MOBILEE1705> if I do this /usr/local/freeswitch/bin/fs_cli -x "expand originate sofia/gateway/\${distributor(pstn_check_number)}/001111111111 loopback/9999999999999" should I have a 9999999999999 extension condition in the dialplan and hold the call to maintain the session alive ? Thanks Franck ----- Original Message ----- From: Steven Ayre To: freeswitch-users at lists.freeswitch.org Sent: Friday, June 04, 2010 3:38 AM Subject: Re: [Freeswitch-users] check status of a number It means the originate command succeeded to create a call, which that UUID identifies. It doesn't tell you anything about the result of that call though - you'll need to watch for events for that UUID, or check its XML CDR to find out the callflow (to see if it matched an extension) or the hangup reason. -Steve On 4 June 2010 04:51, Madovsky wrote: Ok I succeed doing like this : /usr/local/freeswitch/bin/fs_cli -x "expand originate sofia/gateway/\${distributor(pstn_international_2)}/003434REMOTEEXT loopback/999LOCALEXT" and it gives a session: +OK d0c598b5-9291-47eb-a14e-7845763425f4 does +OK means the number always exists ? is there only one ring, hangup and clean the originate call ? Thanks F ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 03, 2010 2:22 PM Subject: Re: [Freeswitch-users] check status of a number Use the loopback endpoint with originate to do this. /b On Jun 3, 2010, at 1:16 PM, Madovsky wrote: Hi, I'd like to use fs_cli to check if the callee number exists and get the status result. Is anyone already did that ? Thanks Franck -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/a3e67744/attachment-0001.html From christian.loeschenkohl at xpirio.com Fri Jun 4 09:08:48 2010 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Fri, 04 Jun 2010 18:08:48 +0200 Subject: [Freeswitch-users] outbound socket problem / fs_ivrd general protection, segfault In-Reply-To: References: <4C08C3E2.3020703@xpirio.com> <4C091026.5040209@xpirio.com> <7F6151C9-4819-471E-B962-552DDB324253@freeswitch.org> <4C091619.9020200@xpirio.com> Message-ID: <4C092510.7040202@xpirio.com> thank you very much updating to the latest version did cure all my problems i will stay on trunk from now on sorry that i didn't try this as first step br On 2010-06-04 17:18, Anthony Minessale wrote: > it's not just about ivrd > > its about the whole esl library and everything that uses it. > > Every time you update FS you must also update all of the ESL and ESL > modules > > 2010/6/4 Christian L?schenkohl > > > hello > > the running version is > http://files.freeswitch.org/freeswitch-1.0.6.tar.bz2 > > i compared the ivrd source with the version i did get with > git clone git://git.freeswitch.org/freeswitch.git > > > br > > On 2010-06-04 16:44, Brian West wrote: > > > Are you using GIT to pull the src? > > > > /b > > > > On Jun 4, 2010, at 9:39 AM, Christian L?schenkohl wrote: > > > >> hi > >> > >> thank you > >> > >> - i use a very up-to-date version - see version output 1.0.6 > >> the ivrd hasn't changed from 1.0.6 to the current trunk > >> > >> - i don't know how to debug with gdb here > >> freeswitch dumps it's core as a core file but fs_ivrd doesn't > >> could you provide a command that does the job? > >> strace > >> > >> also syslog says the kernel logs this > >> Jun 4 14:23:43 vts02 kernel: [2321314.877379] fs_ivrd[18070] > general protection ip:7f2f7b94bbf5 sp:7fffffffa148 error:0 in > libpthread-2.7.so [7f2f7b942000+16000] > >> or > >> Jun 4 14:47:53 vts02 kernel: [2323898.324458] fs_ivrd[19023]: > segfault at a006c6c85 ip 7f0429f43bf5 sp 7fffffffab68 error 4 in > libpthread-2.7.so [7f0429f3a000+16000] > >> > >> fs_ivrd doesn't break and it does not need a restart > >> > >> br > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From kris at kriskinc.com Fri Jun 4 09:04:57 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 4 Jun 2010 12:04:57 -0400 Subject: [Freeswitch-users] Help debugging INVITE with provider In-Reply-To: References: <07314285-F095-4946-B041-D50CD875326D@gmail.com> <404ccbd42e100f414bab82b5bd65461e@thom.fr.eu.org> <8C40FBE2-46D2-4881-83D1-502713FEFABE@freeswitch.org> <066d6f86556d985bcb0d7ad47d2c763c@thom.fr.eu.org> Message-ID: I've been seeing this more and more lately... I had to sign up for a prepaid account from a popular "provider" the other day (don't ask) and the INVITE from them advertised all kinds of crazy codecs I knew their upstreams didn't support. Stuff like G722, G726, AMR, ilbc, etc. Don't get me wrong, these are great codecs in the appropriate scenario but I *KNOW* their upstreams only support G729 and G711. Did they just not bother to change the default codec list or what? Very strange. On Fri, Jun 4, 2010 at 11:13 AM, Brian West wrote: > You can't fix this... the invite is coming from your provider with those in it.. and its invalid so you have no way to correct this unless you install a proxy sever and rewrite the SDP before you send it to freeswitch... your provder's VERY NICE sip proxy isn't so very NICE. > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From frisch.alan at gmail.com Fri Jun 4 09:12:47 2010 From: frisch.alan at gmail.com (Alan Frisch) Date: Fri, 4 Jun 2010 12:12:47 -0400 Subject: [Freeswitch-users] FS "Holds On" to ITSP's 183 for 2 Sec.Before Relaying to Phone? Message-ID: Hi, Just tinkering around with FreeSWITCH pretty much using the default configuration, with some customization to connect my Polycom 430 and my ITSP. The problem I am having is that the first few seconds of calls (or early media if available) have no audio. When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP with timestamps, I can see that FreeSwitch is holding onto the 183 and subsequent 200 OK for about 2 seconds before relaying it onto the Polycom. FWIW, server is an Intel Atom 330. Any idea why this is... I am sure it must be a configuration issue somewhere? I did not have this issue with Asterisk, so I am hoping to troubleshoot it and move on with FreeSWITCH Thanks. AF. From brian at freeswitch.org Fri Jun 4 09:55:25 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Jun 2010 11:55:25 -0500 Subject: [Freeswitch-users] FS "Holds On" to ITSP's 183 for 2 Sec.Before Relaying to Phone? In-Reply-To: References: Message-ID: <63DA6802-CDE6-4632-90FA-A23ACC4490E0@freeswitch.org> Alan, Welcome to FreeSWITCH, Lets start out with some questions... Is FreeSWITCH behind nat? Can you type "sofia status profile internal" and reply with that? /b On Jun 4, 2010, at 11:12 AM, Alan Frisch wrote: > Hi, > > Just tinkering around with FreeSWITCH pretty much using the default > configuration, with some customization to connect my Polycom 430 and > my ITSP. The problem I am having is that the first few seconds of > calls (or early media if available) have no audio. > > When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP > with timestamps, I can see that FreeSwitch is holding onto the 183 and > subsequent 200 OK for about 2 seconds before relaying it onto the > Polycom. FWIW, server is an Intel Atom 330. > > Any idea why this is... I am sure it must be a configuration issue > somewhere? I did not have this issue with Asterisk, so I am hoping to > troubleshoot it and move on with FreeSWITCH > > Thanks. > > AF. From anthony.minessale at gmail.com Fri Jun 4 09:56:44 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Jun 2010 11:56:44 -0500 Subject: [Freeswitch-users] FS "Holds On" to ITSP's 183 for 2 Sec.Before Relaying to Phone? In-Reply-To: References: Message-ID: type the following at the cli console loglevel debug sofia profile internal siptrace on repeat and post the trace to http://pastebin.freeswitch.org (pass is in the dialog box) On Fri, Jun 4, 2010 at 11:12 AM, Alan Frisch wrote: > Hi, > > Just tinkering around with FreeSWITCH pretty much using the default > configuration, with some customization to connect my Polycom 430 and > my ITSP. The problem I am having is that the first few seconds of > calls (or early media if available) have no audio. > > When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP > with timestamps, I can see that FreeSwitch is holding onto the 183 and > subsequent 200 OK for about 2 seconds before relaying it onto the > Polycom. FWIW, server is an Intel Atom 330. > > Any idea why this is... I am sure it must be a configuration issue > somewhere? I did not have this issue with Asterisk, so I am hoping to > troubleshoot it and move on with FreeSWITCH > > Thanks. > > AF. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100604/37fc7593/attachment.html From jcasale at activenetwerx.com Fri Jun 4 10:25:36 2010 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Fri, 4 Jun 2010 17:25:36 +0000 Subject: [Freeswitch-users] Freeswicth nms In-Reply-To: References: <1111639352-1275526834-cardhu_decombobulator_blackberry.rim.net-2040530345-@bda2058.bisx.prodap.on.blackberry> Message-ID: >Are there any FreeSWITCH specific plugins available for Nagios? None that I am aware of, but I use Nagios and how I accomplish this type of need across multiple systems of software and hardware is w/ snmp extends and Nagios check_snmp string regex's. For example, you can extend an snmp command to check hardware raid cards or fs consoles or whatever, then you set the Nagios check_snmp command to expect a regex or specific string in the output. I use this for LSI cards, DRBD and all sorts of things... http://www.logix.cz/michal/devel/nagios/ HTH, jlc From frisch.alan at gmail.com Fri Jun 4 10:28:39 2010 From: frisch.alan at gmail.com (Alan Frisch) Date: Fri, 4 Jun 2010 13:28:39 -0400 Subject: [Freeswitch-users] FS "Holds On" to ITSP's 183 for 2 Sec.Before Relaying to Phone? In-Reply-To: References: Message-ID: Brian/Anthony, Looks like I forgot to remove the <-- and --> from around the custom RTP settings. This may have been causing the hang and calls are now getting full early media, without cut-offs. Thanks and have a great weekend! AF. On Fri, Jun 4, 2010 at 12:56 PM, Anthony Minessale wrote: > type the following at the cli > console loglevel debug > sofia profile internal siptrace on > repeat and post the trace to http://pastebin.freeswitch.org (pass is in the > dialog box) > > On Fri, Jun 4, 2010 at 11:12 AM, Alan Frisch wrote: >> >> Hi, >> >> Just tinkering around with FreeSWITCH pretty much using the default >> configuration, with some customization to connect my Polycom 430 and >> my ITSP. ?The problem I am having is that the first few seconds of >> calls (or early media if available) have no audio. >> >> When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP >> with timestamps, I can see that FreeSwitch is holding onto the 183 and >> subsequent 200 OK for about 2 seconds before relaying it onto the >> Polycom. ?FWIW, server is an Intel Atom 330. >> >> Any idea why this is... I am sure it must be a configuration issue >> somewhere? ?I did not have this issue with Asterisk, so I am hoping to >> troubleshoot it and move on with FreeSWITCH >> >> Thanks. >> >> AF. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 delorenzodesign at gmail.com Fri Jun 4 10:46:43 2010 From: delorenzodesign at gmail.com (Michael De Lorenzo) Date: Fri, 4 Jun 2010 13:46:43 -0400 Subject: [Freeswitch-users] Set Privacy:id on Outbound Call Message-ID: I tried setting the sip_h_Privacy via: freeswitch.Session("{sip_cid_type=pid,origination_caller_id_name=" .. caller_id_name .. ",origination_caller_id_number=+1" .. cid .. ",sip_h_Privacy=id,ignore_early_media=true}sofia/gateway/mygateway/" .. call_this_number) but in the SIP Capture Privacy is still none: INVITE sip:9735551234 at 5.6.7.8 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKSaXt6Bva8UD5K Max-Forwards: 70 From: "Anonymous" ;transport=udp>;tag=QjjNSt2m3g0rB To: > Call-ID: 9b32aa4a-eaa3-122d-9bbb-000bdb94aab9 CSeq: 131718385 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "Anonymous" > v=0 o=FreeSWITCH 1275645559 1275645560 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 27884 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Am I doing something incorrectly? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/aad6f0cd/attachment-0001.html From david.ponzone at gmail.com Fri Jun 4 11:18:14 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Fri, 4 Jun 2010 20:18:14 +0200 Subject: [Freeswitch-users] Issue with Privacy: id and IVR Message-ID: Hello all, I noticed a weird issue when receiving a call with Privacy: id on an IVR. Privacy: id is present on leg B, although I dont set it manually. On a regular DID (bridged directly to leg B without IVR), I would have to set Privacy: id manually. It's like it's forwarded from leg A automagically. My relevant dialplan is the following: (that's the first extension in the dialplan) The IVR config is: I am running latest git. Behaviour is the same on a 1 month old version. Am I missing something ? Thank you David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/f9bcc33a/attachment.html From bwibowo at gmail.com Fri Jun 4 14:50:50 2010 From: bwibowo at gmail.com (Budi wibowo) Date: Fri, 4 Jun 2010 21:50:50 +0000 Subject: [Freeswitch-users] Freeswicth nms In-Reply-To: References: <1111639352-1275526834-cardhu_decombobulator_blackberry.rim.net-2040530345-@bda2058.bisx.prodap.on.blackberry> Message-ID: <786848484-1275688255-cardhu_decombobulator_blackberry.rim.net-2134459084-@bda057.bisx.prodap.on.blackberry> The nms I require can do alarm, performance and statistic. So must be able to read fs specofic oid. Thx -----Original Message----- From: Phillip Jones Sender: freeswitch-users-bounces at lists.freeswitch.org Date: Fri, 4 Jun 2010 11:53:17 To: Reply-To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswicth nms _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 troy at chronostelecom.com Fri Jun 4 16:56:46 2010 From: troy at chronostelecom.com (Troy Anderson) Date: Fri, 4 Jun 2010 16:56:46 -0700 Subject: [Freeswitch-users] Presence/BLF Message-ID: <0F7A39C5-AEBB-42F1-96FF-E92A06387804@chronostelecom.com> I am trying to get my head around presence/BLF indicators in a mixed endpoint environment. We have a couple Cisco 7940's and a couple Aastra phones (55i and 57i), and some Polycoms (450 and others). Presence works great if I am just using the Aastras, but gets wonky when I mix in other phones. I am using Wireshark to try to understand what's going on, but need some advise on what to look for. Is all this handled with SIP NOTIFY events? Is FS completely in charge of the BLF status on the phones? Thanks for any pointers! -Troy From brian at freeswitch.org Fri Jun 4 17:08:12 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Jun 2010 19:08:12 -0500 Subject: [Freeswitch-users] Presence/BLF In-Reply-To: <0F7A39C5-AEBB-42F1-96FF-E92A06387804@chronostelecom.com> References: <0F7A39C5-AEBB-42F1-96FF-E92A06387804@chronostelecom.com> Message-ID: <8AD10A35-4E19-42D0-BF2C-680C8822D245@freeswitch.org> You will not be able to accomplish your goals with a 7940 involved... and you can't accomplish SLA with the Aastra involved. And you can't mix types of phones unless they all do SLA properly. Aastra and Snom do not... Cisco 79XX's do not do BLF/SLA so you're SOL. /b On Jun 4, 2010, at 6:56 PM, Troy Anderson wrote: > I am trying to get my head around presence/BLF indicators in a mixed endpoint environment. We have a couple Cisco 7940's and a couple Aastra phones (55i and 57i), and some Polycoms (450 and others). Presence works great if I am just using the Aastras, but gets wonky when I mix in other phones. > > I am using Wireshark to try to understand what's going on, but need some advise on what to look for. Is all this handled with SIP NOTIFY events? Is FS completely in charge of the BLF status on the phones? > > Thanks for any pointers! > > -Troy From mark.maly at molcs.org Fri Jun 4 17:30:14 2010 From: mark.maly at molcs.org (Mark Maly) Date: Fri, 4 Jun 2010 19:30:14 -0500 Subject: [Freeswitch-users] Presence/BLF In-Reply-To: <8AD10A35-4E19-42D0-BF2C-680C8822D245@freeswitch.org> References: <0F7A39C5-AEBB-42F1-96FF-E92A06387804@chronostelecom.com> <8AD10A35-4E19-42D0-BF2C-680C8822D245@freeswitch.org> Message-ID: <003401cb0446$44a9ed80$cdfdc880$@maly@molcs.org> After the May firmware update, I think I've had pretty good success w/Aastra, but I'm a novice at this. SLA/SCA seem to work now, but I don't know the technical/behind the scenes aspects. I have a 53i and 2 6731i's. That's 'bout all I know... Mark -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 04, 2010 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Presence/BLF You will not be able to accomplish your goals with a 7940 involved... and you can't accomplish SLA with the Aastra involved. And you can't mix types of phones unless they all do SLA properly. Aastra and Snom do not... Cisco 79XX's do not do BLF/SLA so you're SOL. /b On Jun 4, 2010, at 6:56 PM, Troy Anderson wrote: > I am trying to get my head around presence/BLF indicators in a mixed endpoint environment. We have a couple Cisco 7940's and a couple Aastra phones (55i and 57i), and some Polycoms (450 and others). Presence works great if I am just using the Aastras, but gets wonky when I mix in other phones. > > I am using Wireshark to try to understand what's going on, but need some advise on what to look for. Is all this handled with SIP NOTIFY events? Is FS completely in charge of the BLF status on the phones? > > Thanks for any pointers! > > -Troy _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 Fri Jun 4 18:02:26 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Jun 2010 20:02:26 -0500 Subject: [Freeswitch-users] Set Privacy:id on Outbound Call In-Reply-To: References: Message-ID: {cid_type=none,sip_h_Privacy=id} you have to disable any other privacy headers or it will not let you add Privacy (which is think is deprecated anyway) On Fri, Jun 4, 2010 at 12:46 PM, Michael De Lorenzo < delorenzodesign at gmail.com> wrote: > I tried setting the sip_h_Privacy via: > > freeswitch.Session("{sip_cid_type=pid,origination_caller_id_name=" .. > caller_id_name .. ",origination_caller_id_number=+1" .. cid .. > ",sip_h_Privacy=id,ignore_early_media=true}sofia/gateway/mygateway/" .. > call_this_number) > > but in the SIP Capture Privacy is still none: > > INVITE sip:9735551234 at 5.6.7.8 SIP/2.0 > Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKSaXt6Bva8UD5K > Max-Forwards: 70 > From: "Anonymous" > ;transport=udp>;tag=QjjNSt2m3g0rB > To: > > Call-ID: 9b32aa4a-eaa3-122d-9bbb-000bdb94aab9 > CSeq: 131718385 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 > 15:06:05 -0400 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, refer > Privacy: none > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 291 > X-FS-Support: update_display > P-Asserted-Identity: "Anonymous" > > > > v=0 > o=FreeSWITCH 1275645559 1275645560 IN IP4 1.2.3.4 > s=FreeSWITCH > c=IN IP4 1.2.3.4 > t=0 0 > m=audio 27884 RTP/AVP 0 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > Am I doing something incorrectly? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100604/d6c7356a/attachment-0001.html From djbinter at gmail.com Fri Jun 4 20:15:10 2010 From: djbinter at gmail.com (DJB INTERNATIONAL) Date: Fri, 4 Jun 2010 20:15:10 -0700 Subject: [Freeswitch-users] Question regarding table channels in core.db Message-ID: I would like to know whether this is the normal behavior when i ran a query during the call for table channels: select * from channels I noticed that the ip_addr for inbound and outbound shows the same value. Should it show the bleg_ip_addr for outbound direction instead? git-0152706 2010-06-04 19-37-04 -0500 Thank you, Dorn B. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100604/95470ee2/attachment.html From bwibowo at gmail.com Fri Jun 4 20:29:31 2010 From: bwibowo at gmail.com (Budi wibowo) Date: Sat, 5 Jun 2010 03:29:31 +0000 Subject: [Freeswitch-users] Db auth Message-ID: <1831563157-1275708573-cardhu_decombobulator_blackberry.rim.net-1570988617-@bda057.bisx.prodap.on.blackberry> Dear all Anybody knows how to do authentication based on mysql db? For easier maintenance I want to move the profile to database instead of file config. Any idea or turorial are highly appreciated and welcome. /b Budi From shaikbashaatc at yahoo.com Fri Jun 4 22:32:06 2010 From: shaikbashaatc at yahoo.com (Shaik basha) Date: Fri, 4 Jun 2010 22:32:06 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch download and configuration Setup webpage Message-ID: <479303.50690.qm@web43403.mail.sp1.yahoo.com> Dear all, Good morning every one. I a m a new bie in freeswitch, though I tried to search the download page and configuration set up. but, I don't see any where. Can any one help me in this regard. I have spent several hours, though I was not succeeded. Hence I kindly request to let me know from where I can download and how to do configuration setup. Thanking in advance. earliest response in this regard would be very much appreciated. I would be very thankful and grateful for your kind information. Regards, shaikbashaatc +919246769086 From wasim at convergence.pk Fri Jun 4 22:43:03 2010 From: wasim at convergence.pk (Wasim Baig) Date: Sat, 5 Jun 2010 10:43:03 +0500 Subject: [Freeswitch-users] Freeswitch download and configuration Setup webpage In-Reply-To: <479303.50690.qm@web43403.mail.sp1.yahoo.com> References: <479303.50690.qm@web43403.mail.sp1.yahoo.com> Message-ID: http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ http://wiki.freeswitch.org/wiki/Download_FreeSWITCH http://wiki.freeswitch.org/wiki/Installation_Guide -waism On Sat, Jun 5, 2010 at 10:32, Shaik basha wrote: > > Dear all, > > Good morning every one. I a m a new bie in freeswitch, though I tried to > search the download page and configuration set up. but, I don't see any > where. Can any one help me in this regard. I have spent several hours, > though I was not succeeded. > > Hence I kindly request to let me know from where I can download and how to > do configuration setup. Thanking in advance. earliest response in this > regard would be very much appreciated. I would be very thankful and grateful > for your kind information. Regards, > > shaikbashaatc > +919246769086 > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100605/5c12c3a6/attachment.html From vfclists at gmail.com Fri Jun 4 23:27:07 2010 From: vfclists at gmail.com (Frank Church) Date: Sat, 5 Jun 2010 07:27:07 +0100 Subject: [Freeswitch-users] Can freeswitch work with ODBC for Microsoft Access files? In-Reply-To: References: Message-ID: Hasn't anyone tried at all? Is the JET Engine that hated? On 4 June 2010 08:55, Frank Church wrote: > > I'd just like to know if any brave souls have tested Freeswitch with > Microsoft ODBC for JET 4.0 using Microsoft Access or any of the other file > formats it supports, like DBF and Paradox db files. > > Anyone? > > -- > Frank Church > > ======================= > http://devblog.brahmancreations.com > -- Frank Church ======================= http://devblog.brahmancreations.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100605/bb9c9754/attachment.html From david.ponzone at gmail.com Sat Jun 5 06:03:56 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Sat, 5 Jun 2010 15:03:56 +0200 Subject: [Freeswitch-users] Issue with Privacy: id and IVR In-Reply-To: References: Message-ID: <49B8C1B6-5FD3-4F99-A8E1-914A7D7AC4E0@gmail.com> Thanks to Anthony, I got the answer. In fact, the issue had nothing to do with IVR. It's just that I was not setting sip_cid_type to none before bridging from my IVR. I was absolutely not aware about the interactions between Privacy and sip_cid_type. I am now. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 04/06/2010 ? 20:18, David Ponzone a ?crit : > Hello all, > > I noticed a weird issue when receiving a call with Privacy: id on an > IVR. > Privacy: id is present on leg B, although I dont set it manually. > On a regular DID (bridged directly to leg B without IVR), I would > have to set Privacy: id manually. > It's like it's forwarded from leg A automagically. > > My relevant dialplan is the following: > > > > > > > > (that's the first extension in the dialplan) > > The IVR config is: > > greet-long="/usr/local/freeswitch/sounds/vera" > timeout="50000" > inter-digit-timeout="2000" > max-failures="3" > max-timeouts="3" > digit-len="1"> > param="execute_extension set:ringback=/usr/local/freeswitch/sounds/ > trying > ,bridge:sofia/gateway/mosaica/0174031897 inline"/> > > > > I am running latest git. > Behaviour is the same on a 1 month old version. > > Am I missing something ? > > Thank you > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, > d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur. > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100605/1cf60868/attachment-0001.html From asannucci at gmail.com Sat Jun 5 06:26:13 2010 From: asannucci at gmail.com (bakko) Date: Sat, 5 Jun 2010 15:26:13 +0200 Subject: [Freeswitch-users] Skypopen problem Message-ID: Hi, I'm trying tu install skypopen on my centOS server witout success. This is my configuration: virtual xen server on linode.com centos 5.5 32 bit freswitch latest trunk xen-devel given from linode If i try to compile the alsa-driver-1.0.20 modifing the files and lines indicates in the wiki when i do make this is the result: /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c: In function 'snd_pcm_oss_open_file' /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: 'fmode_t' undeclared (first use in this function) /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: (Each undeclared identifier is reported only once /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: for each function it appears in.) If i try to compile the alsa-driver-1.0.20 without any modification and then compile dummy.c, when i try to load snd-dummy module this is the result modprobe snd-dummy WARNING: Error inserting snd_page_alloc (/lib/modules/2.6.18.8-linode22/misc/acore/snd-page-alloc.ko): Invalid module format WARNING: Error inserting snd_timer (/lib/modules/2.6.18.8-linode22/misc/acore/snd-timer.ko): Invalid module format WARNING: Error inserting snd_pcm (/lib/modules/2.6.18.8-linode22/misc/acore/snd-pcm.ko): Invalid module format FATAL: Error inserting snd_dummy (/lib/modules/2.6.18.8-linode22/misc/drivers/snd-dummy.ko): Invalid module format modprobe -f snd-dummy WARNING: Error inserting snd_page_alloc (/lib/modules/2.6.18.8-linode22/misc/acore/snd-page-alloc.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.18.8-linode22/misc/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.18.8-linode22/misc/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_dummy (/lib/modules/2.6.18.8-linode22/misc/drivers/snd-dummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) Any idea? BR - Andrea Sannucci - From jmesquita at freeswitch.org Sat Jun 5 07:41:03 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 5 Jun 2010 11:41:03 -0300 Subject: [Freeswitch-users] Db auth In-Reply-To: <1831563157-1275708573-cardhu_decombobulator_blackberry.rim.net-1570988617-@bda057.bisx.prodap.on.blackberry> References: <1831563157-1275708573-cardhu_decombobulator_blackberry.rim.net-1570988617-@bda057.bisx.prodap.on.blackberry> Message-ID: You have to use mod_xml_curl or mod_xml_odbc. http://wiki.freeswitch.org/wiki/Mod_xml_odbc http://wiki.freeswitch.org/wiki/Mod_xml_curl Haven't tried the latter but curl works like a peach. JM On Sat, Jun 5, 2010 at 12:29 AM, Budi wibowo wrote: > Dear all > Anybody knows how to do authentication based on mysql db? > For easier maintenance I want to move the profile to database instead of > file config. > Any idea or turorial are highly appreciated and welcome. > > /b > Budi > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100605/25f1474d/attachment.html From mike at jerris.com Sat Jun 5 10:54:24 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 5 Jun 2010 13:54:24 -0400 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> References: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> Message-ID: <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> Why would it be an advantage to have your media proxies use another protocol? Mike On Jun 4, 2010, at 1:59 AM, David Ponzone wrote: > It doesn't solve the issue that all the media servers will do signaling too, and will talk SIP with the carriers. > So the carriers will need to allow all the media servers . > > The only clean solution to avoid that, I think, is to have signaling boxes allocating resources from media servers with another protocol than SIP. > RTPproxy does that I think, but I am not sure how it works. > > David Ponzone -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100605/ec5ad187/attachment.html From nazim.aghabayov at gmail.com Sat Jun 5 11:01:56 2010 From: nazim.aghabayov at gmail.com (Nazim Aghabayov) Date: Sat, 05 Jun 2010 23:01:56 +0500 Subject: [Freeswitch-users] Freeswicth nms In-Reply-To: <86ACEC07BA3045CB9EAE360A3A8F4E07@MOBILEE1705> References: <895033401-1275521270-cardhu_decombobulator_blackberry.rim.net-328548762-@bda057.bisx.prodap.on.blackberry><1275579239.1789.7.camel@lenovo400> <86ACEC07BA3045CB9EAE360A3A8F4E07@MOBILEE1705> Message-ID: <4C0A9114.6020203@gmail.com> You may try zabbix ) On 06/04/2010 12:15 AM, Madovsky wrote: > also ganglia... > > ----- Original Message ----- > From: "bakko" > To: > Sent: Thursday, June 03, 2010 3:02 PM > Subject: Re: [Freeswitch-users] Freeswicth nms > > > >> I'm using monit... >> >> :) >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > From tayeb.meftah at gmail.com Sun Jun 6 11:34:09 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sun, 06 Jun 2010 20:34:09 +0200 Subject: [Freeswitch-users] get your job aguinst freeswitch Message-ID: <4C0BEA21.3000008@gmail.com> hello list, i am meftah tayeb, a blind person from algeria that was using freeswitch sunse 1.0.1 release i started firstly with asterisk, that was the very strangett voip application in my life and thank to miconda that redirected me to freeswitch, from asterisk in #openser in 2008 i started learning voip basic in dec 2008 thank to the #freeswitch folk that teached me all this, including firstly anthm, Michael S Collin, brian (BKW), sekil the nice GUI and all other in ogust 2009, algeria telecom and the algeria gouvernmant started blocking sip traffic i was not using it for voip business, but, honestly, just to connect to the public freeswitch conference and the weekely voip users conference i start a complain in my local city, no reply from AT i decided to go to the general office, because i can't use my PC without SIP i got the general directore secrutary ok, he receyved me and heare me saying why you are blocking sip? so he asked me why you need sip? do you do voip business without autorisation? i explaned to him my actual situation and he say: ok, no problem i will open the sip for you, but conditionaly the next mondey you will be here i say ok no problem so, i returned to my home and i see something new: 1. a static ip address linked to my ADSL account 2. sip completly open ok, next monday i was in the general office and i meet the gebneral directotore surprisingly he asked me: do you have a job? i say no but he say yes, you have, tel me i say no aguin and he say you work for algeria telecom ;) he surprised me with this now, he gave me a: good job free house free care with driver so please say thank to the freeswitch project especialy the owner and try to donate to him a much a pocible thank you -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz From jan.berger at video24.no Sat Jun 5 13:01:00 2010 From: jan.berger at video24.no (Jan Berger) Date: Sat, 5 Jun 2010 22:01:00 +0200 Subject: [Freeswitch-users] get your job aguinst freeswitch In-Reply-To: <4C0BEA21.3000008@gmail.com> References: <4C0BEA21.3000008@gmail.com> Message-ID: Nice story Tayeb, good luck with your new job. Jan -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Meftah Tayeb Sent: 6. juni 2010 20:34 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] get your job aguinst freeswitch hello list, i am meftah tayeb, a blind person from algeria that was using freeswitch sunse 1.0.1 release i started firstly with asterisk, that was the very strangett voip application in my life and thank to miconda that redirected me to freeswitch, from asterisk in #openser in 2008 i started learning voip basic in dec 2008 thank to the #freeswitch folk that teached me all this, including firstly anthm, Michael S Collin, brian (BKW), sekil the nice GUI and all other in ogust 2009, algeria telecom and the algeria gouvernmant started blocking sip traffic i was not using it for voip business, but, honestly, just to connect to the public freeswitch conference and the weekely voip users conference i start a complain in my local city, no reply from AT i decided to go to the general office, because i can't use my PC without SIP i got the general directore secrutary ok, he receyved me and heare me saying why you are blocking sip? so he asked me why you need sip? do you do voip business without autorisation? i explaned to him my actual situation and he say: ok, no problem i will open the sip for you, but conditionaly the next mondey you will be here i say ok no problem so, i returned to my home and i see something new: 1. a static ip address linked to my ADSL account 2. sip completly open ok, next monday i was in the general office and i meet the gebneral directotore surprisingly he asked me: do you have a job? i say no but he say yes, you have, tel me i say no aguin and he say you work for algeria telecom ;) he surprised me with this now, he gave me a: good job free house free care with driver so please say thank to the freeswitch project especialy the owner and try to donate to him a much a pocible thank you -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 moises.silva at gmail.com Sat Jun 5 16:11:26 2010 From: moises.silva at gmail.com (Moises Silva) Date: Sat, 5 Jun 2010 19:11:26 -0400 Subject: [Freeswitch-users] get your job aguinst freeswitch In-Reply-To: References: <4C0BEA21.3000008@gmail.com> Message-ID: Wow, very nice story ... Telephony also changed my life, from working as an intern in a micro voip startup in Guadalajara, Mexico writing PHP AGI scripts to work happily now in Sangoma hacking on open source telephony every day :-) Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Sat, Jun 5, 2010 at 4:01 PM, Jan Berger wrote: > Nice story Tayeb, good luck with your new job. > > Jan > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Meftah > Tayeb > Sent: 6. juni 2010 20:34 > To: FreeSWITCH-users at lists.freeswitch.org > Subject: [Freeswitch-users] get your job aguinst freeswitch > > hello list, > i am meftah tayeb, a blind person from algeria that was using freeswitch > sunse 1.0.1 release > i started firstly with asterisk, that was the very strangett voip > application in my life > and thank to miconda that redirected me to freeswitch, from asterisk in > #openser in 2008 > i started learning voip basic in dec 2008 > thank to the #freeswitch folk that teached me all this, including > firstly anthm, Michael S Collin, brian (BKW), sekil the nice GUI and all > other > in ogust 2009, algeria telecom and the algeria gouvernmant started > blocking sip traffic > i was not using it for voip business, but, honestly, just to connect to > the public freeswitch conference and the weekely voip users conference > i start a complain in my local city, no reply from AT > i decided to go to the general office, because i can't use my PC without > SIP > i got the general directore secrutary > ok, he receyved me and heare me saying why you are blocking sip? > so he asked me > why you need sip? > do you do voip business without autorisation? > i explaned to him my actual situation and he say: ok, no problem i will > open the sip for you, but conditionaly > the next mondey you will be here > i say ok no problem > so, i returned to my home and i see something new: > 1. a static ip address linked to my ADSL account > 2. sip completly open > ok, next monday i was in the general office and i meet the gebneral > directotore surprisingly > he asked me: > do you have a job? > i say no > but he say yes, you have, tel me > i say no aguin > and he say you work for algeria telecom > ;) > he surprised me with this > now, he gave me a: > good job > free house > free care with driver > so please say thank to the freeswitch project especialy the owner and > try to donate to him a much a pocible > thank you > > -- > Meftah Tayeb > alg?rie t?l?com SPA > phone: +21321761805 > phone (INUM): +883510001289101 > mobile : +213660347746 > mobile (INUM: +883510001289110 > http://www.algerietelecom.dz > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100605/48516e95/attachment-0001.html From gcd at i.ph Sat Jun 5 19:22:27 2010 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 6 Jun 2010 10:22:27 +0800 Subject: [Freeswitch-users] get your job aguinst freeswitch In-Reply-To: References: <4C0BEA21.3000008@gmail.com> Message-ID: Wow again meftah! It's an inspiration for us. From the Philippines - nandy On Sun, Jun 6, 2010 at 7:11 AM, Moises Silva wrote: > Wow, very nice story ... > > Telephony also changed my life, from working as an intern in a micro voip > startup in Guadalajara, Mexico writing PHP AGI scripts to work happily now > in Sangoma hacking on open source telephony every day :-) > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > > > On Sat, Jun 5, 2010 at 4:01 PM, Jan Berger wrote: > >> Nice story Tayeb, good luck with your new job. >> >> Jan >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Meftah >> Tayeb >> Sent: 6. juni 2010 20:34 >> To: FreeSWITCH-users at lists.freeswitch.org >> Subject: [Freeswitch-users] get your job aguinst freeswitch >> >> hello list, >> i am meftah tayeb, a blind person from algeria that was using freeswitch >> sunse 1.0.1 release >> i started firstly with asterisk, that was the very strangett voip >> application in my life >> and thank to miconda that redirected me to freeswitch, from asterisk in >> #openser in 2008 >> i started learning voip basic in dec 2008 >> thank to the #freeswitch folk that teached me all this, including >> firstly anthm, Michael S Collin, brian (BKW), sekil the nice GUI and all >> other >> in ogust 2009, algeria telecom and the algeria gouvernmant started >> blocking sip traffic >> i was not using it for voip business, but, honestly, just to connect to >> the public freeswitch conference and the weekely voip users conference >> i start a complain in my local city, no reply from AT >> i decided to go to the general office, because i can't use my PC without >> SIP >> i got the general directore secrutary >> ok, he receyved me and heare me saying why you are blocking sip? >> so he asked me >> why you need sip? >> do you do voip business without autorisation? >> i explaned to him my actual situation and he say: ok, no problem i will >> open the sip for you, but conditionaly >> the next mondey you will be here >> i say ok no problem >> so, i returned to my home and i see something new: >> 1. a static ip address linked to my ADSL account >> 2. sip completly open >> ok, next monday i was in the general office and i meet the gebneral >> directotore surprisingly >> he asked me: >> do you have a job? >> i say no >> but he say yes, you have, tel me >> i say no aguin >> and he say you work for algeria telecom >> ;) >> he surprised me with this >> now, he gave me a: >> good job >> free house >> free care with driver >> so please say thank to the freeswitch project especialy the owner and >> try to donate to him a much a pocible >> thank you >> >> -- >> Meftah Tayeb >> alg?rie t?l?com SPA >> phone: +21321761805 >> phone (INUM): +883510001289101 >> mobile : +213660347746 >> mobile (INUM: +883510001289110 >> http://www.algerietelecom.dz >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100606/71a46ce7/attachment.html From infos at madovsky.org Sat Jun 5 19:42:29 2010 From: infos at madovsky.org (Madovsky) Date: Sat, 5 Jun 2010 22:42:29 -0400 Subject: [Freeswitch-users] get your job aguinst freeswitch References: <4C0BEA21.3000008@gmail.com> Message-ID: <0C0EEBDF393C45078E784F16ED7A2A71@MOBILEE1705> great story Meftah good luck with your new job Franck ----- Original Message ----- From: Moises Silva To: freeswitch-users at lists.freeswitch.org Sent: Saturday, June 05, 2010 7:11 PM Subject: Re: [Freeswitch-users] get your job aguinst freeswitch Wow, very nice story ... Telephony also changed my life, from working as an intern in a micro voip startup in Guadalajara, Mexico writing PHP AGI scripts to work happily now in Sangoma hacking on open source telephony every day :-) Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Sat, Jun 5, 2010 at 4:01 PM, Jan Berger wrote: Nice story Tayeb, good luck with your new job. Jan -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Meftah Tayeb Sent: 6. juni 2010 20:34 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] get your job aguinst freeswitch hello list, i am meftah tayeb, a blind person from algeria that was using freeswitch sunse 1.0.1 release i started firstly with asterisk, that was the very strangett voip application in my life and thank to miconda that redirected me to freeswitch, from asterisk in #openser in 2008 i started learning voip basic in dec 2008 thank to the #freeswitch folk that teached me all this, including firstly anthm, Michael S Collin, brian (BKW), sekil the nice GUI and all other in ogust 2009, algeria telecom and the algeria gouvernmant started blocking sip traffic i was not using it for voip business, but, honestly, just to connect to the public freeswitch conference and the weekely voip users conference i start a complain in my local city, no reply from AT i decided to go to the general office, because i can't use my PC without SIP i got the general directore secrutary ok, he receyved me and heare me saying why you are blocking sip? so he asked me why you need sip? do you do voip business without autorisation? i explaned to him my actual situation and he say: ok, no problem i will open the sip for you, but conditionaly the next mondey you will be here i say ok no problem so, i returned to my home and i see something new: 1. a static ip address linked to my ADSL account 2. sip completly open ok, next monday i was in the general office and i meet the gebneral directotore surprisingly he asked me: do you have a job? i say no but he say yes, you have, tel me i say no aguin and he say you work for algeria telecom ;) he surprised me with this now, he gave me a: good job free house free care with driver so please say thank to the freeswitch project especialy the owner and try to donate to him a much a pocible thank you -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100605/59640929/attachment.html From infos at madovsky.org Sat Jun 5 19:46:09 2010 From: infos at madovsky.org (Madovsky) Date: Sat, 5 Jun 2010 22:46:09 -0400 Subject: [Freeswitch-users] simple fees bill Message-ID: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> Hi, any advice for a simple fees module ? NibbleBill seems to be interesting, but maybe there are others. I m only focused on call unit fees, no CDR or call details. Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100605/33b93fd1/attachment.html From dujinfang at gmail.com Sat Jun 5 19:57:04 2010 From: dujinfang at gmail.com (Seven Du) Date: Sun, 6 Jun 2010 10:57:04 +0800 Subject: [Freeswitch-users] get your job aguinst freeswitch In-Reply-To: <0C0EEBDF393C45078E784F16ED7A2A71@MOBILEE1705> References: <4C0BEA21.3000008@gmail.com> <0C0EEBDF393C45078E784F16ED7A2A71@MOBILEE1705> Message-ID: Great ! 2010/6/6 Madovsky : > great story Meftah > good luck with your new job > > Franck > > ----- Original Message ----- > From: Moises Silva > To: freeswitch-users at lists.freeswitch.org > Sent: Saturday, June 05, 2010 7:11 PM > Subject: Re: [Freeswitch-users] get your job aguinst freeswitch > Wow, very nice story ... > Telephony also changed my life, from working as an intern in a micro voip > startup in Guadalajara, Mexico writing PHP AGI scripts to work happily now > in Sangoma hacking on open source telephony every day :-) > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 > Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > > On Sat, Jun 5, 2010 at 4:01 PM, Jan Berger wrote: >> >> Nice story Tayeb, good luck with your new job. >> >> Jan >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Meftah >> Tayeb >> Sent: 6. juni 2010 20:34 >> To: FreeSWITCH-users at lists.freeswitch.org >> Subject: [Freeswitch-users] get your job aguinst freeswitch >> >> hello list, >> i am meftah tayeb, a blind person from algeria that was using freeswitch >> sunse 1.0.1 release >> i started firstly with asterisk, that was the very strangett voip >> application in my life >> and thank to miconda that redirected me to freeswitch, from asterisk in >> #openser in 2008 >> i started learning voip basic in dec 2008 >> thank to the #freeswitch folk that teached me all this, including >> firstly anthm, Michael S Collin, brian (BKW), sekil the nice GUI and all >> other >> in ogust 2009, algeria telecom and the algeria gouvernmant started >> blocking sip traffic >> i was not using it for voip business, but, honestly, just to connect to >> the public freeswitch conference and the weekely ?voip users conference >> i start a complain in my local city, no reply from AT >> i decided to go to the general office, because i can't use my PC without >> SIP >> i got the general directore secrutary >> ok, he receyved me and heare me saying why you are blocking sip? >> so he asked me >> why you need sip? >> do you do voip business without autorisation? >> i explaned to him my actual situation and he say: ok, no problem i will >> open the sip for you, but conditionaly >> the next mondey you will be here >> i say ok no problem >> so, i returned to my home and i see something new: >> 1. a static ip address linked to my ADSL account >> 2. sip completly open >> ok, next monday i was in the general office and i meet the gebneral >> directotore surprisingly >> he asked me: >> do you have a job? >> i say no >> but he say yes, you have, tel me >> i say no aguin >> and he say you work for algeria telecom >> ;) >> he surprised me with this >> now, he gave me a: >> good job >> free house >> free care with driver >> so please say thank to the freeswitch project especialy the owner and >> try to donate to him a much a pocible >> thank you >> >> -- >> Meftah Tayeb >> alg?rie t?l?com SPA >> phone: +21321761805 >> phone (INUM): +883510001289101 >> mobile : +213660347746 >> mobile (INUM: +883510001289110 >> http://www.algerietelecom.dz >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > ________________________________ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn From tayeb.meftah at gmail.com Sun Jun 6 23:17:50 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Mon, 07 Jun 2010 08:17:50 +0200 Subject: [Freeswitch-users] simple fees bill In-Reply-To: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> References: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> Message-ID: <4C0C8F0E.6040000@gmail.com> hello list, i think there is only mod_nibblebill developed by the core FreePBX-V3 developer except of mod_cdr/mod_xml_cdr/all cdr's modules Madovsky, any nibblebill problem? i would be glad to help you thanks Le 06/06/2010 04:46, Madovsky a ?crit : > Hi, > any advice for a simple fees module ? > NibbleBill seems to be interesting, but maybe there are > others. I m only focused on call unit fees, no CDR or > call details. > Thanks > Franck > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/ca46f9ff/attachment.html From babak.freeswitch at gmail.com Sun Jun 6 02:41:13 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Sun, 6 Jun 2010 14:11:13 +0430 Subject: [Freeswitch-users] odbc on windows Message-ID: Hi I configured my sofia profile internal to use odbc. it connects and creates tables but after that I get lots of errors like bellow these are just a few of them: 2010-06-06 14:09:04.000000 [ERR] switch_odbc.c:427 ERR: [BEGIN] [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Inc orrect syntax near 'BEGIN'. ] 2010-06-06 14:09:04.000000 [ERR] switch_core_sqldb.c:404 SQL ERR [STATE: 42000 C ODE 8180 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Statement(s) coul d not be prepared. ] thanx all -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100606/d3d8f770/attachment.html From saeedahmad1981 at gmail.com Sun Jun 6 06:49:35 2010 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Sun, 6 Jun 2010 15:49:35 +0200 Subject: [Freeswitch-users] get your job aguinst freeswitch In-Reply-To: References: <4C0BEA21.3000008@gmail.com> <0C0EEBDF393C45078E784F16ED7A2A71@MOBILEE1705> Message-ID: Great! All the best with new job. On Jun 6, 2010 5:02 AM, "Seven Du" wrote: Great ! 2010/6/6 Madovsky : > great story Meftah > good luck with your new job > > Franck > > ----- Original Message ----- > Fro... -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at list... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100606/78e1fbd2/attachment.html From tayeb.meftah at gmail.com Mon Jun 7 06:46:20 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Mon, 07 Jun 2010 15:46:20 +0200 Subject: [Freeswitch-users] Fwd: [Kde-accessibility] OpenTTS 0.1 released. Message-ID: <4C0CF82C.6040102@gmail.com> -------- Message original -------- Sujet: [Kde-accessibility] OpenTTS 0.1 released. Date : Sun, 6 Jun 2010 13:13:40 +1000 De : Luke Yelavich Pour : kde-accessibility at kde.org I am proud to announce the very first release of OpenTTS, version 0.1. Thanks to everybody who made this release possible, you know who you are. The first release series is focused on delivering a text to speech framework for the GNOME orca screen reader, that is the same, and hopefully better, than the deprecated gnome-speech solution that has been in use ever since the GNOME accessibility project's inception. The OpenTTS development team is also planning on making further usability improvements, which will aid both system administrators, and end users alike. PLEASE NOTE: This release is NOT considered stable. It is recommended that only those who are willing to test, and report bugs use this first release of OpenTTS. Since we are aiming to provide a usable solution for orca, our aim is to deliver a stable release at the time GNOME 3 officially ships. Between now and September therefore, we hope to make regular monthly releases for the community to track our work, test, and give feedback. These releases will not necessarily be considered stable, but will give users a snapshot they can use, which is free of major show stopper bugs. Release notes: * OpenTTS can be used as a system wide service, however at the moment, this is slightly extra work. The OpenTTS development team hopes to make OpenTTS as a system service much easier to use in the near future. Where can I get it? OpenTTS 0.1 is available from the following locations: Tarball: http://files.opentts.org/opentts/opentts-0.1.tar.gz git: git://git.opentts.org/opentts.git Getting involved: We are always willing to accept any help, no matter where it comes from, or what skills you bring to the table. If you would like to contact us and get involved, please do so in the following ways: * OpenTTS project mailing lists, at http://lists.opentts.org. Please sign up to the opentts-dev mailing list for development discussions. Moderate traffic, low signal to noise ratio. There is also opentts-users, for user questinos, and support. Extremely low traffic, currently low signal to noise ratio. * Reporting/triaging bugs on the OpenTTS issue tracker, available at http://www.opentts.org. Bugs can also be reported on the mailing lists, if you prefer. We welcome any and all contributions. Changelog: For those interested, below you will find a shortlog of work that has been done since the project was started just over 2 months ago: Andrei Kholodnyi (46): moved spd_audio_endian to AudioID moved log_level from spd_audio.h to spd_audio.c and rename it moved pulse.c from spd_audio.c to Makefile.am moved alsa.c from spd_audio.c to Makefile.am moved oss.c from spd_audio.c to Makefile.am moved libao.c from spd_audio.c to Makefile.am moved nas.c from spd_audio.c to Makefile.am moved AudioID alloc into plugins move pulse specific data from AudioID to pulse.c move nas specific data from AudioID to nas.c move oss specific data from AudioID to oss.c move alsa specific data from AudioID to alsa.c remove AudioOutputType split spd library and spd plugin headers added plugin name to audio plugin structure move audio static plugins initialization into static_plugins.c moved module_main.c from the dummy.c to Makefile.am moved module_main.c from cicero.c to Makefile.am moved module_main.c from espeak.c to Makefile.am moved module_main.c from festival.c to Makefile.am moved module.main.c from flite.c to Makefile.am moved module_main.c from generic.c to Makefile.am moved module_main.c from ibmtts.c to Makefile.am moved module_main.c from ivona.c to Makefile.am moved fdsetconv.c from module_utils.c to Makefile.am added pkgconfig support moved debug defines to libspeechd.c update pkg-config support added dynamic load of audio plugins fix module_dbgfile memory leak fix rep_line memory leak replace EPunctMode with SPDPunctuation replace ECapLetRecogn with SPDCapitalLetters replace ESpellMode with SPDSpelling replace EVoiceType with SPDVoiceType replace VoiceDescription with SPDVoice replace ENotification with SPDNotification replace int ssml_mode with SPDDataMode replace-int-priority-with-SPDPriority moved EMessageType to SPDMessageType in opentts_types.h moved SPDMsgSettings from fdset.h to module_utils.h fix refs to fdset.h, general text review fix p_client_socket memory leak fix o_buf memory leak fix o_buf memory_leaks in parse.c fix make distcheck Christopher Brannon (113): Prevent automake from failing when we have no ChangeLog. Added spd_audio_get_playcmd function. Removed SPDAudioSettings Add missing include directives. Use const * consistently for function tables. Fix memory leaks in module_audio_init_spd. Fix copyright oversight. Give Andrei credit in the AUTHORS file. Add prototypes for module_utils_addvoice.c. Add a prototype for festival_read_response. Fix calls to ivona_get_msgpart. Declare functions from ivona_client.c in a header file. Possibly include sndfile.h in ivona_client.c. Fix inclusion of config.h for two files. Add ivona_client.h to Makefile.am. Remove use of strdup in do_list_voices. Change LIST VOICES in the internal module protocol. Use glib consistently in command handlers. Don't redefine PACKAGE and VERSION. Reorganized options.c and options.h for the "say" client. Cleanup options.c and options.h for the server. Stop using xfree in the audio subsystem. Fix link order in the modules. Use ltdl to handle static plugins. Use glib allocation in the server. Use glib allocation in the modules. Use glib allocation in the audio subsystem. Use glib allocation in the common library. Surround calls to OL_RET with braces, and add semicolons. Remove some complex macros. Simplify a macro in parse.c. Indent remaining files. Properly indent header files. Replace calls to free with g_free. Fix segfault when compiled against libao 1.0.0. Remove redundant library from modules/Makefile.am. Add new code for generating timestamps. Rename the Python bindings, including the speechd_config suite. Fix a mistake in the python subdirectory. Use automake in the session subdirectory. Convert the python subdirectory to use the autotools. spd_audio_endian should be id->format. Use otts_getline instead of getline. Fix module_add_config_option in module_utils.c. Change the way that lines are read in the server. Do not include the common library in libopentts. Reopen the connection to pulseaudio when the audio format changes. Add an extra newline when writing to .profile. Add the Unix socket communication method, and use it by default. The communication method is now configurable. Support Unix sockets in the C client API. Autospawn is controlled by an argument to spd_open2. Make autospawn optional for clients. The daemon no longer checks SPEECHD_PORT. Don't require the ~/.speech-dispatcher directory to already exist. Autospawn mechanism fine-tuning Redirect stderr when autospawning in the Python library. Check for the existence of the pid file before handling autospawn. Update otts-conf to generate config for Unix sockets. Do not require ~/.speech-dispatcher to exist. Remove some unused options from the configure script. Put Unix sockets in the home directory, and respect SPEECHD_SOCK. Don't try to compile OSS support if OSS is not available. Fix compilation on Mac OS X. Correct the loop boundary in a convenience function. Properly handle the return value of spd_sayf in the otts-say client. Improved portability checks. Don't store pointers to strings that dotconf will free. Move spd_audio.c and spd_audio.h to the modules subdirectory. Fix prefixes of entry functions in the audio modules. Adjust init_settings_tables to account for renaming of constants. Add constants for reporting errors. Install opentts_types.h. More renaming. Remove the "Future Design" section. More replacement. Change pathnames in the documentation. Rename speech-dispatcher.texi to opentts.texi. Let the otts-say client speak messages that begin with dash. Additional renaming in the otts-say client. Change names of files. Fix a bug in libopentts.c. Replace many references to Speech Dispatcher. Fix two memory leaks in the parse_set function. Use proper comparisons while traversing the history list. Prevent another macro from storing pointers to dotconf's strings. Fix: two command-line options had no effect. Fix a nasty off-by-one bug in the Festival module. Bugfix: ALSA now resumes when suspended. Fix an error-handling clause in alsa_play. Change the test-runner program to use Unix sockets. Update the documentation to reflect the code. Replace the SPEECHD_* environment variables with OPENTTSD_*. Add an introduction to the C API section of the manual. Only change endianness for 16-bit audio. Set correct endianness during libao playback. The function flite_set_voice should be a no-op. Discuss our history in the documentation. Fix the help and version messages for the server. Change speechd to openttsd in log messages. Change speechd to openttsd in error messages and comments. Amend authorship and copyright info for the docs. otts-say.texi needs more authorship and copyright info. Cast return value of lt_dlsym to the proper type. Update the AUTHORS file. Document the process of building ibmtts for 64-bit. Unknown voice types should map to SPD_NO_VOICE. Refactor spd_open2 and fix spd_close. Properly handle autospawning in the Python library. Break out of a loop in pulse_play when an error occurs. Wait for the auto-spawned server process to terminate. Use the same parameter to listen() for both Unix and TCP sockets. Break out of a loop in libao_play when an error occurs. Jason White (1): fix otts-conf default port Jos? Vilmar Est?cio de Souza (1): memory leak Luke Yelavich (15): Initial OpenTTSd git repository creation based on unofficial 0.6.68 release of speech-dispatcher session - Re-add accidentally lost Makefile.in Re-add lost python build file Remove reference to ChangeLog file in top directory Reset version to 0.0 Re-add yet another lost file from the openttsd repo transition Set correct audio endianness for audio playback using pulseaudio server - change last references to speech-dispatcher config file directories to opentts Change all other ~/.speech-dispatcher references to ~/.opentts Change more speech-dispatcher references in the pulseaudio output code Rename speechd/spd-say/Speech Dispatcher to their OpenTTS equivalents in otts-conf Don't refer to non-existant fdset.h file in src/common Distribute fdset.h in the tarball Link against libcommon.la in the build dir, not the src dir Release 0.1 Rui Batista (1): Renamed Speech Dispatcher naming to Opentts in INSTALL and minor cleanups on that file Steve Holmes (2): Updated documentation to reflect OpenTTS name. Updated opentts.desktop file. Trevor Saunders (24): change calls to signal to sigaction clean up module_utils update docs rename variables and functions in the server rename spdsend to otts-send cleanup documentation for otts-send rename files in src/clients/send/ change references to spdsend to otts-send update comment in libao.c rename spd_audio_log_level to audio_log_level renamed several audio types rename some audio functions change LIBSPEECHD_DEBUG macro to LIBOPENTTS_DEBUG rename SPEECHD_DEBUG to OPENTTSD_DEBUG and move it to openttsd.h rename SpeechdSocket to openttsd_sockets renamed multiple functions update several comments rename options.home_speechd_dir to options.opentts_dir rename SPEECHD_OPTION_xxx macros change the include guard in openttsd.h to use openttsd instead of speechd change TSpeechDQueue to queue_t rename TSpeechDSock to sock_t rename speechd_queue_alloc to queue_alloc rename SPEECHD_DEFAULT_PORT to OPENTTSD_DEFAULT_PORT William Hubbs (83): rename configure.in to configure.ac fixed typo updated extra_dist in main makefile update documentation Makefile rearranged the api and client directories fix issues with installation only build test binaries when make check is run clean up several makefiles distribute the API's for common lisp and guile started cleaning up configure.ac detect sound systems before speech synthesizers define installed headers correctly don't declare libraries in LDFLAGS more makefile cleanup fixed an autotools conflict include config.h in all sources move the code in intl to a convenience library tests should use the convenience library remove PACKAGE and VERSION definitions from server's makefile clean up ignore patterns cleaned up EXTRA_DIST for modules added more ignore patterns convert libsdaudio to a convenience library configuration system update remove rule for html documentation fixed a comment remove old speech dispatcher announcement disable session integration support by default clean up the modules makefile removed extra_dist from src/audio/Makefile.am install spd_audio_plugin header file rename the C API library tests should link against libopentts fix include in the say client fixed an include in the guile bindings updated the header in libopentts.h updated an include in libopentts.c rename directories from speech-dispatcher to opentts audio subsystem updates move the modules sub directory rename ottslibdir to audiodir fixed variable in audio makefile include spd_audio_plugin.h from spd_audio.c instead of spd_audio.h. Revert "include spd_audio_plugin.h from spd_audio.c instead of spd_audio.h." ran code through Lindent python build updates session build fixes do not install Czech documentation fix declaration of festival_connection_crashed fix include in spd_audio.c fix declaration of current_index_mark convert global variable declarations to externs consolidate global variables in modules remove DBG macro from modules remove the prefix from the module names rename the daemon make better use of conditionals in modules makefile rename fixes move public header files to common include directory build system updates remove the session directory add configure option to disable the installation of the python bindings remove prefix from audio plugin names Revert "remove prefix from audio plugin names" removed several change logs remove prefix from names of audio plugins fix includes of libopentts.h and opentts_types.h in the say client and api change main config file to openttsd.conf fix config makefile remove duplicate include directive in openttsd.conf move desktop autostart file to config directory clean up config files fix permissions on the socket permission fixes fix typo remove dotconf from pkg-config file fix Makefile substitutions clarify the contact information in the SSIP document fix copyright assignment request make configure fail when necessary libraries are not on the system build fixes for the audio subsystem remove ability to disable autospawn from server fix implicit declaration of g_unlink jose vilmar estacio de souza (1): possible memory leaks tbsaunde (1): refactor module_main to use disatch_cmd instead of macros Regards Luke Yelavich OpenTTS Project lead. _______________________________________________ Opentts-dev mailing list Opentts-dev at lists.opentts.org http://lists.opentts.org/listinfo.cgi/opentts-dev-opentts.org _______________________________________________ kde-accessibility mailing list kde-accessibility at kde.org https://mail.kde.org/mailman/listinfo/kde-accessibility -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/e0b9a0ee/attachment-0001.html From freeswitch at peely.com Sun Jun 6 08:17:24 2010 From: freeswitch at peely.com (peely) Date: Sun, 6 Jun 2010 08:17:24 -0700 (PDT) Subject: [Freeswitch-users] uuid_media hangs Message-ID: <1275837444425-5145657.post@n2.nabble.com> Hi, I'm trying to use the uuid_media command to control the media path after a call has been established. I can't seem to get it to respond wither through the console or through ESL. Whenever I issue a uuid_media {uuid} or uuid_media off {uuid} the console or connection just hangs, even when I have debug output I get nothing. I've tried this with both bridged and originated calls. I have enable-3pcc set to true on the profile. Effectively I am trying to set up an IVR through ESL then connect the caller on to a destination, once connected I want to push the media away. Looking at the SIP coming in from the far end I see invite and update is allowed so in theory this should be supported by the remote party. Could somebody please let me know what the conditions are to be able to use uuid_media? Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5145657.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Sun Jun 6 08:58:04 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 6 Jun 2010 17:58:04 +0200 Subject: [Freeswitch-users] Ang.: uuid_media hangs Message-ID: <3E8F7997-4E17-4F1D-9840-760CC1E01AA1@visionutveckling.se> Update to latest git, Tony has fixed tjusiga... Peter Olsson ----- Reply message ----- Fr?n: "peely" Datum: s?n, jun 6, 2010 17:24 Rubrik: [Freeswitch-users] uuid_media hangs Till: "freeswitch-users at lists.freeswitch.org" Hi, I'm trying to use the uuid_media command to control the media path after a call has been established. I can't seem to get it to respond wither through the console or through ESL. Whenever I issue a uuid_media {uuid} or uuid_media off {uuid} the console or connection just hangs, even when I have debug output I get nothing. I've tried this with both bridged and originated calls. I have enable-3pcc set to true on the profile. Effectively I am trying to set up an IVR through ESL then connect the caller on to a destination, once connected I want to push the media away. Looking at the SIP coming in from the far end I see invite and update is allowed so in theory this should be supported by the remote party. Could somebody please let me know what the conditions are to be able to use uuid_media? Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5145657.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4c0bbda332931523817751! From brian at freeswitch.org Sun Jun 6 10:37:20 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 6 Jun 2010 12:37:20 -0500 Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1275837444425-5145657.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com> Message-ID: update this was already fixed. /b Sent from my iPad On Jun 6, 2010, at 10:17 AM, peely wrote: > > Hi, > > I'm trying to use the uuid_media command to control the media path after a > call has been established. I can't seem to get it to respond wither through > the console or through ESL. > > Whenever I issue a uuid_media {uuid} or uuid_media off {uuid} the console or > connection just hangs, even when I have debug output I get nothing. > > I've tried this with both bridged and originated calls. I have enable-3pcc > set to true on the profile. > > Effectively I am trying to set up an IVR through ESL then connect the caller > on to a destination, once connected I want to push the media away. > > Looking at the SIP coming in from the far end I see invite and update is > allowed so in theory this should be supported by the remote party. > > Could somebody please let me know what the conditions are to be able to use > uuid_media? > > > > Thanks, > > > Neil. > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5145657.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 freeswitch at peely.com Sun Jun 6 12:14:05 2010 From: freeswitch at peely.com (peely) Date: Sun, 6 Jun 2010 12:14:05 -0700 (PDT) Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1275837444425-5145657.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com> Message-ID: <1275851645813-5146342.post@n2.nabble.com> Thanks. Do you know in which build it was fixed? I pull from the freeswitch nightly ppa on ubuntu so will try to compile the latest snapshot and test. Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5146342.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Sun Jun 6 12:50:57 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 6 Jun 2010 21:50:57 +0200 Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1275851645813-5146342.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com>, <1275851645813-5146342.post@n2.nabble.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C567CA60732@cooper> 4a4670a71ea43306c944ebe87828f105fcc59451 from git /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för peely [freeswitch at peely.com] Skickat: den 6 juni 2010 21:14 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] uuid_media hangs Thanks. Do you know in which build it was fixed? I pull from the freeswitch nightly ppa on ubuntu so will try to compile the latest snapshot and test. Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5146342.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4c0bf51f32932727217645! From freeswitch at peely.com Sun Jun 6 13:58:31 2010 From: freeswitch at peely.com (peely) Date: Sun, 6 Jun 2010 13:58:31 -0700 (PDT) Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1275837444425-5145657.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com> Message-ID: <1275857911630-5146647.post@n2.nabble.com> Just compiled the latest git snapshot, same problem. uuid_media {uuid} hangs. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5146647.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at peely.com Sun Jun 6 14:22:56 2010 From: freeswitch at peely.com (peely) Date: Sun, 6 Jun 2010 14:22:56 -0700 (PDT) Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1275857911630-5146647.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> Message-ID: <1275859376110-5146705.post@n2.nabble.com> Hmm, this seems more due to the latest snapshot having broken background events in esl. I get notification that the filter is added, but no events! If I do a straight Bridge in the dialplan things are a little different. uuid_media {uuid} waits about 5 seconds then ditches the call. A Wireshark trace shows that the reinvite is ent, the remote party then sends an RTP packet to the correct port, but fs is not ready and a Destination Unreachable ICM packet is responded, which makes the remote party clear back the call with a 420. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5146705.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jan.berger at video24.no Sun Jun 6 14:34:25 2010 From: jan.berger at video24.no (Jan Berger) Date: Sun, 6 Jun 2010 23:34:25 +0200 Subject: [Freeswitch-users] ccxml/vxml apps needed for testing Message-ID: Hi, If someone has ccxml and/or vxml apps they are willing to contribute for testing please send them to me off-list. Please include info about what platform they are tested on. The app's will only be used for conformance testing - currently only towards a debug-simulator. jan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100606/d1508e6f/attachment.html From brian at freeswitch.org Sun Jun 6 14:42:52 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 6 Jun 2010 16:42:52 -0500 Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1275859376110-5146705.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> <1275859376110-5146705.post@n2.nabble.com> Message-ID: <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> if you are not pulling direct from our git repo then I'm not sure you have the fixed version yet. please pull from our git repo only. I know first hand this is fixed, I spent hours with Anthony testing it over and over again once the fix was in place... unless you're doing something we did not. /b Sent from my iPad On Jun 6, 2010, at 4:22 PM, peely wrote: > > Hmm, this seems more due to the latest snapshot having broken background > events in esl. I get notification that the filter is added, but no events! > > If I do a straight Bridge in the dialplan things are a little different. > uuid_media {uuid} waits about 5 seconds then ditches the call. A Wireshark > trace shows that the reinvite is ent, the remote party then sends an RTP > packet to the correct port, but fs is not ready and a Destination > Unreachable ICM packet is responded, which makes the remote party clear back > the call with a 420. > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5146705.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 freeswitch at peely.com Sun Jun 6 15:09:35 2010 From: freeswitch at peely.com (peely) Date: Sun, 6 Jun 2010 15:09:35 -0700 (PDT) Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> <1275859376110-5146705.post@n2.nabble.com> <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> Message-ID: <1275862175540-5146803.post@n2.nabble.com> Hi Brian, I am pulling from git and do see the difference. I don't know what's going on with ESL and background events which is causing me some testing headaches but I think I know what's happening with the reinvite. The 420 that comes back seems to be moaning about the Require: timer. header that's passed by FS. The original dialogue negotiated session timers, so FS passes a header in the reinvite requiring timers. It seems that the SBC at the other end doesn't like this and in the 420 a field Unsupported: timer is included. Is there a possibility to have a configuration option which removes the Require: timer in reinvites? Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5146803.html Sent from the freeswitch-users mailing list archive at Nabble.com. From djbinter at gmail.com Sun Jun 6 17:15:56 2010 From: djbinter at gmail.com (DJB International) Date: Sun, 6 Jun 2010 17:15:56 -0700 Subject: [Freeswitch-users] Event did not return anything after update. Message-ID: After updating from 170404a to 9f73ddd, I could not see anything in event message like before. I connect to mod_event_socket by perl script below: require ESL; my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); $con->events("plain", "all"); while($con->connected()) { #my $e = $con->recvEventTimed(100); my $e = $con->recvEvent(); if ($e) { #print $e->serialize(); my $h = $e->firstHeader(); print "\n--------------------RECEIVED--------------------\n"; while ($h) { printf "Header: [%s] = [%s]\n", $h, $e->getHeader($h); $h = $e->nextHeader(); } } } I wonder whether there is any changes that I need to adjust after the update. Thank you, Dorn B. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100606/18516c39/attachment.html From darkeagle6 at hotmail.com Sat Jun 5 13:19:48 2010 From: darkeagle6 at hotmail.com (Myron Curtis) Date: Sat, 5 Jun 2010 13:19:48 -0700 Subject: [Freeswitch-users] speech recognition? Message-ID: Hi, I am developing a virtual world called Virtual Worlds Grid, and I am using FreeSwitch to provide voice. I would like to integrate it with a speech recognition engine like Simon, in order to have everything said in voice to be typed into text. This is one way to make the grid accessible for people who cannot use voice, and it also allows language translation software like Open Bablefish, to translate voice conversations from the resulting text files, which would facilitate multilingual conversations. I am new to the FreeSwitch system, and need some guidance on how to set this up. I would appreciate any tips, cautions, or links to practical help I can get. If this will work in a virtual world, there is no reason it cannot be made to work on cell phones. Thanks, Myron Curtis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100605/863f4898/attachment.html From stephen at stephenjc.com Sun Jun 6 11:55:48 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Sun, 6 Jun 2010 14:55:48 -0400 Subject: [Freeswitch-users] javascript bridge Message-ID: I have the following code, and after the bridge the javascript seems to stop. outsession = new Session("{ringback=\'%(2000,4000,440.0,480.0)\',instant_ringback=true,ignore_early_media=false}sofia/gateway/" + providerhash["providername0"] + "/XXXXXXXXXXXX"); bridge(session,outsession); while(outsession.ready()) { console_log("notice","ping"); } I am looking to manage the b leg, should i use execute_on_answer instead? or is there a way to make the code continue after the bridge. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100606/1b1bf95c/attachment.html From diego.viola at gmail.com Sat Jun 5 20:06:59 2010 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 5 Jun 2010 23:06:59 -0400 Subject: [Freeswitch-users] simple fees bill In-Reply-To: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> References: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> Message-ID: Hi Madovsky, Do you need a pre-paid or post-paid system? Diego On Sat, Jun 5, 2010 at 10:46 PM, Madovsky wrote: > Hi, > > any advice for a simple fees module ? > NibbleBill seems to be interesting, but maybe there are > others. I m only focused on call unit fees, no CDR or > call details. > > Thanks > > Franck > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100605/b0a32766/attachment.html From engineerzuhairraza at gmail.com Sun Jun 6 09:52:33 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Sun, 6 Jun 2010 21:52:33 +0500 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: > Hi everyone > I am working on skype module,according to wiki I have successfuly installed > freeswitch and skype module with alsa driver. I am facing trouble at > creating clone a directory step, I am using centos 5.4 and skype static > version. First i enter command at my freeswitch box under > mod_skypopen/configs > > gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 > > then I install xauth on server after that i ssh with x forwarding to my > server from an another linux desktop and open skype by typing > /usr/bin/skype, it launched skype client at Linux desktop but it didn't ask > for connecting with skypopen api, although it creates ".Skype" directory on > my server but when i load mod_skypopen it says could not find any skype > instance and when i typed skypopen_auth it also says no skype instance found > on X 0:0. Can anyone tell me where I have mistaken?? > > I also observed it connected sometimes but sometimes it does not. > -- > Regards, > Zuhair Raza > > > -- Regards, Zuhair Raza -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100606/ed0fbe19/attachment.html From jonpauldavies at gmail.com Sun Jun 6 13:43:34 2010 From: jonpauldavies at gmail.com (Jon Davies) Date: Sun, 6 Jun 2010 21:43:34 +0100 Subject: [Freeswitch-users] Detect events other than DTMF during .streamFile Message-ID: Hi I was wondering if it was possible to detect events other than DTMF that are fired during .streamFile?. DTMF events are received by my callback without issue, but if I try to use speech detection via mod_pocketsphinx the callback doesn't get fired. If this is not possible, are there any other options available to me in order to detect speech input during the playback of a file? Kind Regards jdee From infos at madovsky.org Sun Jun 6 18:46:52 2010 From: infos at madovsky.org (Madovsky) Date: Sun, 6 Jun 2010 21:46:52 -0400 Subject: [Freeswitch-users] simple fees bill References: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> Message-ID: <1C010464354E40E68B5CC4D34165ABC6@MOBILEE1705> prepaid only. Thanks F ----- Original Message ----- From: Diego Viola To: freeswitch-users at lists.freeswitch.org Sent: Saturday, June 05, 2010 11:06 PM Subject: Re: [Freeswitch-users] simple fees bill Hi Madovsky, Do you need a pre-paid or post-paid system? Diego On Sat, Jun 5, 2010 at 10:46 PM, Madovsky wrote: Hi, any advice for a simple fees module ? NibbleBill seems to be interesting, but maybe there are others. I m only focused on call unit fees, no CDR or call details. Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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/20100606/fb7f7ca4/attachment-0001.html From bwibowo at gmail.com Sun Jun 6 18:51:36 2010 From: bwibowo at gmail.com (Budi wibowo) Date: Mon, 7 Jun 2010 01:51:36 +0000 Subject: [Freeswitch-users] simple fees bill In-Reply-To: References: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> Message-ID: <1915997343-1275875496-cardhu_decombobulator_blackberry.rim.net-298651853-@bda057.bisx.prodap.on.blackberry> Anybody running nibble bill on centos 4? I try to install it but always failed in supporting libraries. -----Original Message----- From: Diego Viola Sender: freeswitch-users-bounces at lists.freeswitch.org Date: Sat, 5 Jun 2010 23:06:59 To: Reply-To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] simple fees bill _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 Sun Jun 6 19:29:16 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 6 Jun 2010 22:29:16 -0400 Subject: [Freeswitch-users] What happen when leg-a & leg-b are on SIP/TCP and one get a socket disconnection during a call? In-Reply-To: <4C050DFD.8080708@infosecurity.ch> References: <4C050DFD.8080708@infosecurity.ch> Message-ID: <462835DE-EAF0-4070-9A6B-373EF139A252@jerris.com> Try turning on tport debugging up to the highest level and see what happens. I can't recall what the sip specs say for this. Mike On Jun 1, 2010, at 9:41 AM, Fabio Pietrosanti (naif) wrote: > Hi, > > sorry for using the so big subject, i am experiencing some strange > behaviour i am going to enter more in depth debugging in the next weeks. > > I have the following situation: > - A call getting established > - both peers are using SIP/TLS over TCP > - leg-b get disconnected at TCP level (such as WiFi disconnection or > just a TCP hard reset by a NAT device reloading it's rules) > > In such condition i noticed, but still need detailed investigation, that > FS does not detect that leg-b TCP socket died, thus notifying > immediately leg-a that the call cannot be completed. > FS instead, it seems from log/tcpdump observation, wait until the > various SIP timer expire in order to provide back to leg-a error that > the call cannot be completed. > > If it's that way, it would be a nice improvement, when a SIP/TCP or > SIP/TLS client get a TCP disconnect, to immediately react on the other > leg of the call by notifying the proper SIP error, because the layer4 > (SIP) connection is lost due to a layer3 (TCP) connection break. > > Is this something feasible/reasonable? From mike at jerris.com Sun Jun 6 19:30:11 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 6 Jun 2010 22:30:11 -0400 Subject: [Freeswitch-users] LUA - ** - Disconnect the call or stop call progress [ ringing or slient] and jump to the next statement In-Reply-To: References: Message-ID: <06BD1937-9C49-4C69-ABCF-B8BB84223290@jerris.com> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app On Jun 1, 2010, at 3:05 PM, Aloysius Lloyd wrote: > Hi All, > > I would like to implement a feature code ** - Disconnect the leg B or stop call progress [ ringing or slient] and jump to the next statement. > > Here is the Lua code I am using to bridge two calls. How to implement the feature code like ** for this purpose. > > session:preAnswer(); > > digits = session:playAndGetDigits(10, 20, 3, 5000, "#", "enter-dest.wav", "invalid-digits.wav", "\\d+|\\*"); > obSession = freeswitch.Session("sofia/gateway/voipms/"..digits,session) > > if obSession:ready() then > obSession:execute("sched_hangup","+60 alloted_timeout"); > freeswitch.bridge(session, obSession); > end From brian at freeswitch.org Sun Jun 6 19:34:05 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 6 Jun 2010 21:34:05 -0500 Subject: [Freeswitch-users] Event did not return anything after update. In-Reply-To: References: Message-ID: <199F2626-6C10-4CCD-9197-2A8E14395C63@freeswitch.org> did you reinstall ESL.so? Sent from my iPad On Jun 6, 2010, at 7:15 PM, DJB International wrote: > After updating from 170404a to 9f73ddd, I could not see anything in event message like before. > > I connect to mod_event_socket by perl script below: > > require ESL; > > my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); > > $con->events("plain", "all"); > > while($con->connected()) { > #my $e = $con->recvEventTimed(100); > my $e = $con->recvEvent(); > > if ($e) { > #print $e->serialize(); > my $h = $e->firstHeader(); > print "\n--------------------RECEIVED--------------------\n"; > while ($h) { > printf "Header: [%s] = [%s]\n", $h, $e->getHeader($h); > $h = $e->nextHeader(); > } > > } > > } > > I wonder whether there is any changes that I need to adjust after the update. > > Thank you, > Dorn B. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 hitormiz13 at yahoo.com Sun Jun 6 19:47:46 2010 From: hitormiz13 at yahoo.com (ro ra) Date: Mon, 7 Jun 2010 10:47:46 +0800 (SGT) Subject: [Freeswitch-users] Registration Update of Contact List In-Reply-To: <76515BA8-C8D1-4C62-9587-A438E1714A75@freeswitch.org> Message-ID: <178327.43392.qm@web76805.mail.sg1.yahoo.com> so if the first Authentication is: Authorization: Digest username="user1", realm="freeswitch.com", qop=auth, nonce="50300e8e-6df3-11df-95bf-fbb7abfa9b66", opaque="", nc=00000001, cnonce="6f54a149", uri="sip:test.freeswitch.com:5060", response="af68c01d136dbc40fe62341afe125bc6" the next should be? ?Authorization: Digest username="user1", realm="freeswitch.com", qop=auth, nonce="50300e8e-6df3-11df-95bf-fbb7abfa9b66", opaque="", nc=00000002, cnonce="6f54a149", uri="sip:test.freeswitch.com:5060", response="af68c01d136dbc40fe62341afe125bc6" is that correct?... --- On Fri, 6/4/10, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] Registration Update of Contact List To: freeswitch-users at lists.freeswitch.org Date: Friday, June 4, 2010, 10:59 PM It should already do this if your user1 device supports the proper nonce count. /b On Jun 4, 2010, at 4:46 AM, ro ra wrote: Hi, ??? what do i need to change on my configuration files to make? this scenario work: ??? user1?? --------- REGISTER????? ------------> Freeswitch ??? user1? <-------- 401 UnAuthorized -------?? Freeswitch ??? user1 ? --------- REGISTER w/ Auth -----> Freeswitch ??? user1?? <------- 200 OK? ------------------------?? Freeswitch ??? ----------------------------5 secs delay---------------------------------- ??? user1 ----------- REGISTER w/ Auth(Update) ----->? Freeswitch ??? user1 <-------- 200 OK?? -----------------------???? Freeswitch ?? or is this possible with freeswitch...? ??? thanks -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/e8bf8bef/attachment.html From djbinter at gmail.com Sun Jun 6 20:06:56 2010 From: djbinter at gmail.com (DJB International) Date: Sun, 6 Jun 2010 20:06:56 -0700 Subject: [Freeswitch-users] Event did not return anything after update. In-Reply-To: <199F2626-6C10-4CCD-9197-2A8E14395C63@freeswitch.org> References: <199F2626-6C10-4CCD-9197-2A8E14395C63@freeswitch.org> Message-ID: Brian, Can you please advise the procedure in git to revert back the revision? Thank you, Dorn B. On Sun, Jun 6, 2010 at 7:34 PM, Brian West wrote: > did you reinstall ESL.so? > > Sent from my iPad > > On Jun 6, 2010, at 7:15 PM, DJB International wrote: > > > After updating from 170404a to 9f73ddd, I could not see anything in event > message like before. > > > > I connect to mod_event_socket by perl script below: > > > > require ESL; > > > > my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); > > > > $con->events("plain", "all"); > > > > while($con->connected()) { > > #my $e = $con->recvEventTimed(100); > > my $e = $con->recvEvent(); > > > > if ($e) { > > #print $e->serialize(); > > my $h = $e->firstHeader(); > > print "\n--------------------RECEIVED--------------------\n"; > > while ($h) { > > printf "Header: [%s] = [%s]\n", $h, $e->getHeader($h); > > $h = $e->nextHeader(); > > } > > > > } > > > > } > > > > I wonder whether there is any changes that I need to adjust after the > update. > > > > Thank you, > > Dorn B. > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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/20100606/fe1df360/attachment.html From brian at freeswitch.org Sun Jun 6 20:11:44 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 6 Jun 2010 22:11:44 -0500 Subject: [Freeswitch-users] Registration Update of Contact List In-Reply-To: <178327.43392.qm@web76805.mail.sg1.yahoo.com> References: <178327.43392.qm@web76805.mail.sg1.yahoo.com> Message-ID: <869A3CDC-5FA5-496C-9192-D9252F45A346@freeswitch.org> no the response should differ if i recall. Sent from my iPad On Jun 6, 2010, at 9:47 PM, ro ra wrote: > so if the first Authentication is: > > Authorization: Digest username="user1", realm="freeswitch.com", qop=auth, nonce="50300e8e-6df3-11df-95bf-fbb7abfa9b66", opaque="", nc=00000001, cnonce="6f54a149", uri="sip:test.freeswitch.com:5060", response="af68c01d136dbc40fe62341afe125bc6" > > the next should be? > > > Authorization: Digest username="user1", realm="freeswitch.com", qop=auth, nonce="50300e8e-6df3-11df-95bf-fbb7abfa9b66", opaque="", nc=00000002, cnonce="6f54a149", uri="sip:test.freeswitch.com:5060", response="af68c01d136dbc40fe62341afe125bc6" > > is that correct?... > > --- On Fri, 6/4/10, Brian West wrote: > > From: Brian West > Subject: Re: [Freeswitch-users] Registration Update of Contact List > To: freeswitch-users at lists.freeswitch.org > Date: Friday, June 4, 2010, 10:59 PM > > It should already do this if your user1 device supports the proper nonce count. > > /b > > On Jun 4, 2010, at 4:46 AM, ro ra wrote: > >> Hi, >> >> what do i need to change on my configuration files to make this scenario work: >> >> user1 --------- REGISTER ------------> Freeswitch >> user1 <-------- 401 UnAuthorized ------- Freeswitch >> user1 --------- REGISTER w/ Auth -----> Freeswitch >> user1 <------- 200 OK ------------------------ Freeswitch >> ----------------------------5 secs delay---------------------------------- >> user1 ----------- REGISTER w/ Auth(Update) -----> Freeswitch >> user1 <-------- 200 OK ----------------------- Freeswitch >> >> or is this possible with freeswitch...? >> >> thanks > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100606/7064a88f/attachment-0001.html From brian at freeswitch.org Sun Jun 6 20:12:51 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 6 Jun 2010 22:12:51 -0500 Subject: [Freeswitch-users] Event did not return anything after update. In-Reply-To: References: <199F2626-6C10-4CCD-9197-2A8E14395C63@freeswitch.org> Message-ID: did you recompile libesl also? you shouldn't have to revert just update correctly. Sent from my iPad On Jun 6, 2010, at 10:06 PM, DJB International wrote: > Brian, > > Can you please advise the procedure in git to revert back the revision? > > Thank you, > Dorn B. > > On Sun, Jun 6, 2010 at 7:34 PM, Brian West wrote: > did you reinstall ESL.so? > > Sent from my iPad > > On Jun 6, 2010, at 7:15 PM, DJB International wrote: > > > After updating from 170404a to 9f73ddd, I could not see anything in event message like before. > > > > I connect to mod_event_socket by perl script below: > > > > require ESL; > > > > my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); > > > > $con->events("plain", "all"); > > > > while($con->connected()) { > > #my $e = $con->recvEventTimed(100); > > my $e = $con->recvEvent(); > > > > if ($e) { > > #print $e->serialize(); > > my $h = $e->firstHeader(); > > print "\n--------------------RECEIVED--------------------\n"; > > while ($h) { > > printf "Header: [%s] = [%s]\n", $h, $e->getHeader($h); > > $h = $e->nextHeader(); > > } > > > > } > > > > } > > > > I wonder whether there is any changes that I need to adjust after the update. > > > > Thank you, > > Dorn B. > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100606/4889375a/attachment.html From infos at madovsky.org Sun Jun 6 20:17:13 2010 From: infos at madovsky.org (Madovsky) Date: Sun, 6 Jun 2010 23:17:13 -0400 Subject: [Freeswitch-users] simple fees bill References: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> <1915997343-1275875496-cardhu_decombobulator_blackberry.rim.net-298651853-@bda057.bisx.prodap.on.blackberry> Message-ID: Seems to work well on Fedora 10 64bits ----- Original Message ----- From: "Budi wibowo" To: Sent: Sunday, June 06, 2010 9:51 PM Subject: Re: [Freeswitch-users] simple fees bill > Anybody running nibble bill on centos 4? > I try to install it but always failed in supporting libraries. > > -----Original Message----- > From: Diego Viola > Sender: freeswitch-users-bounces at lists.freeswitch.org > Date: Sat, 5 Jun 2010 23:06:59 > To: > Reply-To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] simple fees bill > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 From djbinter at gmail.com Sun Jun 6 20:46:11 2010 From: djbinter at gmail.com (DJB International) Date: Sun, 6 Jun 2010 20:46:11 -0700 Subject: [Freeswitch-users] Event did not return anything after update. In-Reply-To: References: <199F2626-6C10-4CCD-9197-2A8E14395C63@freeswitch.org> Message-ID: Yes, Brian. I tried to recompile it, but it did not work. I even tried the php socket to connect direct: http://wiki.freeswitch.org/wiki/PHP_Event_Socket, and it still did not work. I did work for other servers with 170404a version. I am not sure what else I missed or something might have changed in between the version. Thank you, Dorn B. On Sun, Jun 6, 2010 at 8:12 PM, Brian West wrote: > did you recompile libesl also? you shouldn't have to revert just update > correctly. > > Sent from my iPad > > On Jun 6, 2010, at 10:06 PM, DJB International wrote: > > Brian, > > Can you please advise the procedure in git to revert back the revision? > > Thank you, > Dorn B. > > On Sun, Jun 6, 2010 at 7:34 PM, Brian West < > brian at freeswitch.org> wrote: > >> did you reinstall ESL.so? >> >> Sent from my iPad >> >> On Jun 6, 2010, at 7:15 PM, DJB International < >> djbinter at gmail.com> wrote: >> >> > After updating from 170404a to 9f73ddd, I could not see anything in >> event message like before. >> > >> > I connect to mod_event_socket by perl script below: >> > >> > require ESL; >> > >> > my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); >> > >> > $con->events("plain", "all"); >> > >> > while($con->connected()) { >> > #my $e = $con->recvEventTimed(100); >> > my $e = $con->recvEvent(); >> > >> > if ($e) { >> > #print $e->serialize(); >> > my $h = $e->firstHeader(); >> > print "\n--------------------RECEIVED--------------------\n"; >> > while ($h) { >> > printf "Header: [%s] = [%s]\n", $h, $e->getHeader($h); >> > $h = $e->nextHeader(); >> > } >> > >> > } >> > >> > } >> > >> > I wonder whether there is any changes that I need to adjust after the >> update. >> > >> > Thank you, >> > Dorn B. >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > >> FreeSWITCH-users at lists.freeswitch.org >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.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 >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100606/72340f53/attachment.html From djbinter at gmail.com Sun Jun 6 21:04:14 2010 From: djbinter at gmail.com (DJB International) Date: Sun, 6 Jun 2010 21:04:14 -0700 Subject: [Freeswitch-users] Event did not return anything after update. In-Reply-To: References: <199F2626-6C10-4CCD-9197-2A8E14395C63@freeswitch.org> Message-ID: Brian, I did a further test with just telneting to port 8021 and no output came out as well,. # telnet 127.0.0.1 8021 Trying 127.0.0.1... Connected to localhost.localdomain (127.0.0.1). Escape character is '^]'. Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted events plain ALL Content-Type: command/reply Reply-Text: +OK event listener enabled plain (nothing shown) Thank you, Dorn B. On Sun, Jun 6, 2010 at 8:46 PM, DJB International wrote: > Yes, Brian. I tried to recompile it, but it did not work. I even tried > the php socket to connect direct: > http://wiki.freeswitch.org/wiki/PHP_Event_Socket, and it still did not > work. I did work for other servers with 170404a version. I am not sure > what else I missed or something might have changed in between the version. > > Thank you, > Dorn B. > > > On Sun, Jun 6, 2010 at 8:12 PM, Brian West wrote: > >> did you recompile libesl also? you shouldn't have to revert just update >> correctly. >> >> Sent from my iPad >> >> On Jun 6, 2010, at 10:06 PM, DJB International >> wrote: >> >> Brian, >> >> Can you please advise the procedure in git to revert back the revision? >> >> Thank you, >> Dorn B. >> >> On Sun, Jun 6, 2010 at 7:34 PM, Brian West < >> brian at freeswitch.org> wrote: >> >>> did you reinstall ESL.so? >>> >>> Sent from my iPad >>> >>> On Jun 6, 2010, at 7:15 PM, DJB International < >>> djbinter at gmail.com> wrote: >>> >>> > After updating from 170404a to 9f73ddd, I could not see anything in >>> event message like before. >>> > >>> > I connect to mod_event_socket by perl script below: >>> > >>> > require ESL; >>> > >>> > my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); >>> > >>> > $con->events("plain", "all"); >>> > >>> > while($con->connected()) { >>> > #my $e = $con->recvEventTimed(100); >>> > my $e = $con->recvEvent(); >>> > >>> > if ($e) { >>> > #print $e->serialize(); >>> > my $h = $e->firstHeader(); >>> > print "\n--------------------RECEIVED--------------------\n"; >>> > while ($h) { >>> > printf "Header: [%s] = [%s]\n", $h, $e->getHeader($h); >>> > $h = $e->nextHeader(); >>> > } >>> > >>> > } >>> > >>> > } >>> > >>> > I wonder whether there is any changes that I need to adjust after the >>> update. >>> > >>> > Thank you, >>> > Dorn B. >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > >>> FreeSWITCH-users at lists.freeswitch.org >>> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.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 >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100606/88026279/attachment-0001.html From brian at freeswitch.org Sun Jun 6 21:15:36 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 6 Jun 2010 23:15:36 -0500 Subject: [Freeswitch-users] Event did not return anything after update. In-Reply-To: References: <199F2626-6C10-4CCD-9197-2A8E14395C63@freeswitch.org> Message-ID: <815D6B55-E146-4070-8895-70D56C66D8B2@freeswitch.org> Please open a jira and attach all the info you have collected so far. Thanks, Brian On Jun 6, 2010, at 11:04 PM, DJB International wrote: > Brian, > > I did a further test with just telneting to port 8021 and no output came out as well,. > > # telnet 127.0.0.1 8021 > Trying 127.0.0.1... > Connected to localhost.localdomain (127.0.0.1). > Escape character is '^]'. > Content-Type: auth/request > > auth ClueCon > > Content-Type: command/reply > Reply-Text: +OK accepted > > events plain ALL > > Content-Type: command/reply > Reply-Text: +OK event listener enabled plain > > > (nothing shown) > > Thank you, > Dorn B. From brian at freeswitch.org Sun Jun 6 21:25:59 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 6 Jun 2010 23:25:59 -0500 Subject: [Freeswitch-users] Event did not return anything after update. In-Reply-To: References: <199F2626-6C10-4CCD-9197-2A8E14395C63@freeswitch.org> Message-ID: <5E82599B-ED11-4871-8340-A5F54A2B65D7@freeswitch.org> Never mind the Jira just update in a few moments. /b On Jun 6, 2010, at 11:04 PM, DJB International wrote: > Brian, > > I did a further test with just telneting to port 8021 and no output came out as well,. > > # telnet 127.0.0.1 8021 > Trying 127.0.0.1... > Connected to localhost.localdomain (127.0.0.1). > Escape character is '^]'. > Content-Type: auth/request > > auth ClueCon > > Content-Type: command/reply > Reply-Text: +OK accepted > > events plain ALL > > Content-Type: command/reply > Reply-Text: +OK event listener enabled plain > > > (nothing shown) > > Thank you, > Dorn B. > > > > On Sun, Jun 6, 2010 at 8:46 PM, DJB International wrote: > Yes, Brian. I tried to recompile it, but it did not work. I even tried the php socket to connect direct: http://wiki.freeswitch.org/wiki/PHP_Event_Socket, and it still did not work. I did work for other servers with 170404a version. I am not sure what else I missed or something might have changed in between the version. > > Thank you, > Dorn B. > > > On Sun, Jun 6, 2010 at 8:12 PM, Brian West wrote: > did you recompile libesl also? you shouldn't have to revert just update correctly. > > Sent from my iPad > > On Jun 6, 2010, at 10:06 PM, DJB International wrote: > >> Brian, >> >> Can you please advise the procedure in git to revert back the revision? >> >> Thank you, >> Dorn B. >> >> On Sun, Jun 6, 2010 at 7:34 PM, Brian West wrote: >> did you reinstall ESL.so? >> >> Sent from my iPad >> >> On Jun 6, 2010, at 7:15 PM, DJB International wrote: >> >> > After updating from 170404a to 9f73ddd, I could not see anything in event message like before. >> > >> > I connect to mod_event_socket by perl script below: >> > >> > require ESL; >> > >> > my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); >> > >> > $con->events("plain", "all"); >> > >> > while($con->connected()) { >> > #my $e = $con->recvEventTimed(100); >> > my $e = $con->recvEvent(); >> > >> > if ($e) { >> > #print $e->serialize(); >> > my $h = $e->firstHeader(); >> > print "\n--------------------RECEIVED--------------------\n"; >> > while ($h) { >> > printf "Header: [%s] = [%s]\n", $h, $e->getHeader($h); >> > $h = $e->nextHeader(); >> > } >> > >> > } >> > >> > } >> > >> > I wonder whether there is any changes that I need to adjust after the update. >> > >> > Thank you, >> > Dorn B. >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100606/f7decfad/attachment.html From djbinter at gmail.com Sun Jun 6 21:37:30 2010 From: djbinter at gmail.com (DJB International) Date: Sun, 6 Jun 2010 21:37:30 -0700 Subject: [Freeswitch-users] Event did not return anything after update. In-Reply-To: <5E82599B-ED11-4871-8340-A5F54A2B65D7@freeswitch.org> References: <199F2626-6C10-4CCD-9197-2A8E14395C63@freeswitch.org> <5E82599B-ED11-4871-8340-A5F54A2B65D7@freeswitch.org> Message-ID: Oh, I did it as well: http://jira.freeswitch.org/browse/ESL-39. Should I close it? Thank you, Dorn B. On Sun, Jun 6, 2010 at 9:25 PM, Brian West wrote: > Never mind the Jira just update in a few moments. > > /b > > On Jun 6, 2010, at 11:04 PM, DJB International wrote: > > Brian, > > I did a further test with just telneting to port 8021 and no output came > out as well,. > > # telnet 127.0.0.1 8021 > Trying 127.0.0.1... > Connected to localhost.localdomain (127.0.0.1). > Escape character is '^]'. > Content-Type: auth/request > > auth ClueCon > > Content-Type: command/reply > Reply-Text: +OK accepted > > events plain ALL > > Content-Type: command/reply > Reply-Text: +OK event listener enabled plain > > > (nothing shown) > > Thank you, > Dorn B. > > > > On Sun, Jun 6, 2010 at 8:46 PM, DJB International wrote: > >> Yes, Brian. I tried to recompile it, but it did not work. I even tried >> the php socket to connect direct: >> http://wiki.freeswitch.org/wiki/PHP_Event_Socket, and it still did not >> work. I did work for other servers with 170404a version. I am not sure >> what else I missed or something might have changed in between the version. >> >> Thank you, >> Dorn B. >> >> >> On Sun, Jun 6, 2010 at 8:12 PM, Brian West wrote: >> >>> did you recompile libesl also? you shouldn't have to revert just update >>> correctly. >>> >>> Sent from my iPad >>> >>> On Jun 6, 2010, at 10:06 PM, DJB International >>> wrote: >>> >>> Brian, >>> >>> Can you please advise the procedure in git to revert back the revision? >>> >>> Thank you, >>> Dorn B. >>> >>> On Sun, Jun 6, 2010 at 7:34 PM, Brian West < >>> brian at freeswitch.org> wrote: >>> >>>> did you reinstall ESL.so? >>>> >>>> Sent from my iPad >>>> >>>> On Jun 6, 2010, at 7:15 PM, DJB International < >>>> djbinter at gmail.com> wrote: >>>> >>>> > After updating from 170404a to 9f73ddd, I could not see anything in >>>> event message like before. >>>> > >>>> > I connect to mod_event_socket by perl script below: >>>> > >>>> > require ESL; >>>> > >>>> > my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); >>>> > >>>> > $con->events("plain", "all"); >>>> > >>>> > while($con->connected()) { >>>> > #my $e = $con->recvEventTimed(100); >>>> > my $e = $con->recvEvent(); >>>> > >>>> > if ($e) { >>>> > #print $e->serialize(); >>>> > my $h = $e->firstHeader(); >>>> > print "\n--------------------RECEIVED--------------------\n"; >>>> > while ($h) { >>>> > printf "Header: [%s] = [%s]\n", $h, $e->getHeader($h); >>>> > $h = $e->nextHeader(); >>>> > } >>>> > >>>> > } >>>> > >>>> > } >>>> > >>>> > I wonder whether there is any changes that I need to adjust after the >>>> update. >>>> > >>>> > Thank you, >>>> > Dorn B. >>>> > _______________________________________________ >>>> > FreeSWITCH-users mailing list >>>> > >>>> FreeSWITCH-users at lists.freeswitch.org >>>> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.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 >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >>> >>> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100606/047f9d90/attachment-0001.html From djbinter at gmail.com Sun Jun 6 21:57:25 2010 From: djbinter at gmail.com (DJB International) Date: Sun, 6 Jun 2010 21:57:25 -0700 Subject: [Freeswitch-users] Event did not return anything after update. In-Reply-To: References: <199F2626-6C10-4CCD-9197-2A8E14395C63@freeswitch.org> <5E82599B-ED11-4871-8340-A5F54A2B65D7@freeswitch.org> Message-ID: Thank you, Brian. It is working correctly now on 5cd066df . Regards, Dorn B. On Sun, Jun 6, 2010 at 9:37 PM, DJB International wrote: > Oh, I did it as well: http://jira.freeswitch.org/browse/ESL-39. Should > I close it? > > Thank you, > Dorn B. > > > On Sun, Jun 6, 2010 at 9:25 PM, Brian West wrote: > >> Never mind the Jira just update in a few moments. >> >> /b >> >> On Jun 6, 2010, at 11:04 PM, DJB International wrote: >> >> Brian, >> >> I did a further test with just telneting to port 8021 and no output came >> out as well,. >> >> # telnet 127.0.0.1 8021 >> Trying 127.0.0.1... >> Connected to localhost.localdomain (127.0.0.1). >> Escape character is '^]'. >> Content-Type: auth/request >> >> auth ClueCon >> >> Content-Type: command/reply >> Reply-Text: +OK accepted >> >> events plain ALL >> >> Content-Type: command/reply >> Reply-Text: +OK event listener enabled plain >> >> >> (nothing shown) >> >> Thank you, >> Dorn B. >> >> >> >> On Sun, Jun 6, 2010 at 8:46 PM, DJB International wrote: >> >>> Yes, Brian. I tried to recompile it, but it did not work. I even tried >>> the php socket to connect direct: >>> http://wiki.freeswitch.org/wiki/PHP_Event_Socket, and it still did not >>> work. I did work for other servers with 170404a version. I am not sure >>> what else I missed or something might have changed in between the version. >>> >>> Thank you, >>> Dorn B. >>> >>> >>> On Sun, Jun 6, 2010 at 8:12 PM, Brian West wrote: >>> >>>> did you recompile libesl also? you shouldn't have to revert just update >>>> correctly. >>>> >>>> Sent from my iPad >>>> >>>> On Jun 6, 2010, at 10:06 PM, DJB International >>>> wrote: >>>> >>>> Brian, >>>> >>>> Can you please advise the procedure in git to revert back the revision? >>>> >>>> >>>> Thank you, >>>> Dorn B. >>>> >>>> On Sun, Jun 6, 2010 at 7:34 PM, Brian West < >>>> brian at freeswitch.org> wrote: >>>> >>>>> did you reinstall ESL.so? >>>>> >>>>> Sent from my iPad >>>>> >>>>> On Jun 6, 2010, at 7:15 PM, DJB International < >>>>> djbinter at gmail.com> wrote: >>>>> >>>>> > After updating from 170404a to 9f73ddd, I could not see anything in >>>>> event message like before. >>>>> > >>>>> > I connect to mod_event_socket by perl script below: >>>>> > >>>>> > require ESL; >>>>> > >>>>> > my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); >>>>> > >>>>> > $con->events("plain", "all"); >>>>> > >>>>> > while($con->connected()) { >>>>> > #my $e = $con->recvEventTimed(100); >>>>> > my $e = $con->recvEvent(); >>>>> > >>>>> > if ($e) { >>>>> > #print $e->serialize(); >>>>> > my $h = $e->firstHeader(); >>>>> > print "\n--------------------RECEIVED--------------------\n"; >>>>> > while ($h) { >>>>> > printf "Header: [%s] = [%s]\n", $h, $e->getHeader($h); >>>>> > $h = $e->nextHeader(); >>>>> > } >>>>> > >>>>> > } >>>>> > >>>>> > } >>>>> > >>>>> > I wonder whether there is any changes that I need to adjust after the >>>>> update. >>>>> > >>>>> > Thank you, >>>>> > Dorn B. >>>>> > _______________________________________________ >>>>> > FreeSWITCH-users mailing list >>>>> > >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.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 >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>>> >>>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100606/8219481b/attachment.html From tony.tin at noahmedia.com.hk Sun Jun 6 22:41:29 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Mon, 7 Jun 2010 13:41:29 +0800 Subject: [Freeswitch-users] dynamic dial plan with lua Message-ID: Hi, I'm trying to implement dynamic dial plan, my approach is calling a lua scrip in "default.xml" as below, so I can change the dial plan by changing the lua script at any time. Could any on please give me some advice about my approach and whether there is any limitation of LUA regarding dial plan implemenation. Thanks. Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/a1cbae8e/attachment.html From gmaruzz at celliax.org Mon Jun 7 00:07:05 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 7 Jun 2010 09:07:05 +0200 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza wrote: >> version. First i enter command at my freeswitch box under >> mod_skypopen/configs >> >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 >> >> then I install xauth on server after that i ssh with x forwarding to my >> server from an another linux desktop and open skype by typing >> /usr/bin/skype, it launched skype client at Linux desktop but it didn't ask >> for connecting with skypopen api, although it creates ".Skype" directory on After launching the skype client from the ssh -X session, you have to launch skypopen_auth from the same ssh -X session (giving the X server as an argument), eg: "./skypopen_auth $DISPLAY" >> my server but when i load mod_skypopen it says could not find any skype >> instance and when i typed skypopen_auth it also says no skype instance found >> on X 0:0. Can anyone tell me where I have mistaken?? After having given the auth to the skype client, and closing it so it save that auth, you close the ssh -X session, and launch an X server and a skype client in the server, using the script in the configs directory (as explained in the wiki). After having launched that script, you load mod_skypopen. You MUST edit the script and the skypopen.conf.xml to use your own values for skype username and password. -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at celliax.org Mon Jun 7 00:12:50 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 7 Jun 2010 09:12:50 +0200 Subject: [Freeswitch-users] Skypopen problem In-Reply-To: References: Message-ID: On Sat, Jun 5, 2010 at 3:26 PM, bakko wrote: > I'm trying tu install skypopen on my centOS server witout success. > > This is my configuration: > > virtual xen server on linode.com > > centos 5.5 32 bit > > freswitch latest trunk > > xen-devel given from linode > > If i try to compile the alsa-driver-1.0.20 modifing the files and lines > indicates in the wiki when i do make this is the result: > > /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c: In function > 'snd_pcm_oss_open_file' > /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: 'fmode_t' > undeclared > (first use in this function) > /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: (Each > undeclared identifier is reported only once > /usr/src/alsa-driver-1.0.20/acore/oss/pcm_oss.c:2304: error: for each > function it appears in.) > > If i try to compile the alsa-driver-1.0.20 without any modification and then > compile dummy.c, when i try to load snd-dummy module this is the result > > modprobe snd-dummy > WARNING: Error inserting snd_page_alloc > (/lib/modules/2.6.18.8-linode22/misc/acore/snd-page-alloc.ko): Invalid > module format > WARNING: Error inserting snd_timer > (/lib/modules/2.6.18.8-linode22/misc/acore/snd-timer.ko): Invalid module > format > WARNING: Error inserting snd_pcm > (/lib/modules/2.6.18.8-linode22/misc/acore/snd-pcm.ko): Invalid module > format > FATAL: Error inserting snd_dummy > (/lib/modules/2.6.18.8-linode22/misc/drivers/snd-dummy.ko): Invalid module > format > > modprobe -f snd-dummy > WARNING: Error inserting snd_page_alloc > (/lib/modules/2.6.18.8-linode22/misc/acore/snd-page-alloc.ko): Unknown > symbol in module, or unknown parameter (see dmesg) > WARNING: Error inserting snd_timer > (/lib/modules/2.6.18.8-linode22/misc/acore/snd-timer.ko): Unknown symbol in > module, or unknown parameter (see dmesg) > WARNING: Error inserting snd_pcm > (/lib/modules/2.6.18.8-linode22/misc/acore/snd-pcm.ko): Unknown symbol in > module, or unknown parameter (see dmesg) > FATAL: Error inserting snd_dummy > (/lib/modules/2.6.18.8-linode22/misc/drivers/snd-dummy.ko): Unknown symbol > in module, or unknown parameter (see dmesg) Seems that the linode kernel has been patched in a different way from the stock centos kernel. But seems it snd-dummy compiles fine without the modifications needed by the stock centos kernel. So, if you follow the instructions in the wiki, and at the end "make install" the whole alsa-drivers, it will substitute the ALSA installation. Then, you will have to rmmod all the snd* modules, or - better - just reboot the machine. It will probably work. Let us know if you still have problems. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From freeswitch at peely.com Mon Jun 7 00:39:33 2010 From: freeswitch at peely.com (peely) Date: Mon, 7 Jun 2010 00:39:33 -0700 (PDT) Subject: [Freeswitch-users] Change in behaviour for ESL events? Message-ID: <1275896373992-5147891.post@n2.nabble.com> Hi, In the latest git snapshot I've compiled, event subscriptions within ESL no longer seem to function in the same way. I used to issue "filter Unique-ID {uuid}\n\n" and "filter Other-Leg-Unique-ID {uuid}\n\n" followed by "events plain all\n\n". I did this because "myevents\n\n" would not allow me to subscribe to events for background jobs issued by bgapi, which I do quite a lot. Applying the filter then subscribing to all events seemed the most stable and allowed me to subscribe to additional events should I need them. In the latest snapshot, I don't receive any events through this mechanism. I tried "event text all" as newly suggested on the ESL outbound wiki page, but this transmits a heap of white space to my socket then kills freeswitch! Could somebody please tell me if this is something that is "work in progress" and will ultimately resume old behaviour, or should I be doing something else to monitor events for my uuid and anything I spawn in that session? Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Change-in-behaviour-for-ESL-events-tp5147891p5147891.html Sent from the freeswitch-users mailing list archive at Nabble.com. From engineerzuhairraza at gmail.com Mon Jun 7 03:45:39 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Mon, 7 Jun 2010 15:45:39 +0500 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: Thanks for explanation Sir, One more question, please explain cd /root mount /dev/hda5 /mnt cp /mnt/root/skypeconfig2.tgz ./ tar xzf skypeconfig2.tgz chown -R root.root .Skype According to wiki we haven't created a tgz file before that, but .Skype directory at the server On Mon, Jun 7, 2010 at 12:07 PM, Giovanni Maruzzelli wrote: > On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza > wrote: > >> version. First i enter command at my freeswitch box under > >> mod_skypopen/configs > >> > >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 > >> > >> then I install xauth on server after that i ssh with x forwarding to my > >> server from an another linux desktop and open skype by typing > >> /usr/bin/skype, it launched skype client at Linux desktop but it didn't > ask > >> for connecting with skypopen api, although it creates ".Skype" directory > on > > After launching the skype client from the ssh -X session, you have to > launch skypopen_auth from the same ssh -X session (giving the X server > as an argument), eg: "./skypopen_auth $DISPLAY" > > >> my server but when i load mod_skypopen it says could not find any skype > >> instance and when i typed skypopen_auth it also says no skype instance > found > >> on X 0:0. Can anyone tell me where I have mistaken?? > > After having given the auth to the skype client, and closing it so it > save that auth, you close the ssh -X session, and launch an X server > and a skype client in the server, using the script in the configs > directory (as explained in the wiki). > > After having launched that script, you load mod_skypopen. > > You MUST edit the script and the skypopen.conf.xml to use your own > values for skype username and password. > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/bc449d02/attachment.html From jan.berger at video24.no Mon Jun 7 03:46:15 2010 From: jan.berger at video24.no (Jan Berger) Date: Mon, 7 Jun 2010 12:46:15 +0200 Subject: [Freeswitch-users] speech recognition? In-Reply-To: References: Message-ID: <7588B1B7A1B741A4A2A72C38249FC92D@dell9400> Hi It was someone else who requested a similar natural speech to text service earlier.... It is some ASR examples on wiki - pocketsphinx exist + bridging into commercial versions. What is "Simon" ? Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Myron Curtis Sent: 5. juni 2010 22:20 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] speech recognition? Hi, I am developing a virtual world called Virtual Worlds Grid, and I am using FreeSwitch to provide voice. I would like to integrate it with a speech recognition engine like Simon, in order to have everything said in voice to be typed into text. This is one way to make the grid accessible for people who cannot use voice, and it also allows language translation software like Open Bablefish, to translate voice conversations from the resulting text files, which would facilitate multilingual conversations. I am new to the FreeSwitch system, and need some guidance on how to set this up. I would appreciate any tips, cautions, or links to practical help I can get. If this will work in a virtual world, there is no reason it cannot be made to work on cell phones. Thanks, Myron Curtis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/832141cf/attachment.html From gmaruzz at celliax.org Mon Jun 7 03:53:28 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 7 Jun 2010 12:53:28 +0200 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: On Mon, Jun 7, 2010 at 12:45 PM, Zuhair Raza wrote: > Thanks for explanation Sir, One more question, please explain > > cd /root > mount /dev/hda5 /mnt > cp /mnt/root/skypeconfig2.tgz ./ > tar xzf skypeconfig2.tgz > chown -R root.root .Skype > > According to wiki we haven't created a tgz file before that, but .Skype > directory at the server If you do it with ssh -X (as pre the previous mail), you don't need a Skype config directory from another computer. You created that directory. So, just skip those steps. -giovanni > > > On Mon, Jun 7, 2010 at 12:07 PM, Giovanni Maruzzelli > wrote: >> >> On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza >> wrote: >> >> version. First i enter command at my freeswitch box under >> >> mod_skypopen/configs >> >> >> >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 >> >> >> >> then I install xauth on server after that i ssh with x forwarding to my >> >> server from an another linux desktop and open skype by typing >> >> /usr/bin/skype, it launched skype client at Linux desktop but it didn't >> >> ask >> >> for connecting with skypopen api, although it creates ".Skype" >> >> directory on >> >> After launching the skype client from the ssh -X session, you have to >> launch skypopen_auth from the same ssh -X session (giving the X server >> as an argument), eg: "./skypopen_auth $DISPLAY" >> >> >> my server but when i load mod_skypopen it says could not find any skype >> >> instance and when i typed skypopen_auth it also says no skype instance >> >> found >> >> on X 0:0. Can anyone tell me where I have mistaken?? >> >> After having given the auth to the skype client, and closing it so it >> save that auth, you close the ssh -X session, and launch an X server >> and a skype client in the server, using the script in the configs >> directory (as explained in the wiki). >> >> After having launched that script, you load mod_skypopen. >> >> You MUST edit the script and the skypopen.conf.xml to use your own >> values for skype username and password. >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Regards, > Zuhair Raza > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From gustavo.espeche at upper-soft.com Mon Jun 7 04:15:14 2010 From: gustavo.espeche at upper-soft.com (Gustavo Espeche) Date: Mon, 07 Jun 2010 08:15:14 -0300 Subject: [Freeswitch-users] RTP Proxy Message-ID: <1275909314.2807.0.camel@gustavo-laptop> Hi some one know an rtp proxy that work with freeswitch. Thanks. Gustavo Espeche www.easyipcall.com From engineerzuhairraza at gmail.com Mon Jun 7 04:29:05 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Mon, 7 Jun 2010 16:29:05 +0500 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: ok thanks.. but what about my question that skype didnot ask me to to connect to skype API. and when i do either on ./skypopen_auth :101 or :0 it says cannot open X display On Mon, Jun 7, 2010 at 3:53 PM, Giovanni Maruzzelli wrote: > On Mon, Jun 7, 2010 at 12:45 PM, Zuhair Raza > wrote: > > Thanks for explanation Sir, One more question, please explain > > > > cd /root > > mount /dev/hda5 /mnt > > cp /mnt/root/skypeconfig2.tgz ./ > > tar xzf skypeconfig2.tgz > > chown -R root.root .Skype > > > > According to wiki we haven't created a tgz file before that, but .Skype > > directory at the server > > If you do it with ssh -X (as pre the previous mail), you don't need a > Skype config directory from another computer. You created that > directory. > So, just skip those steps. > > -giovanni > > > > > > > On Mon, Jun 7, 2010 at 12:07 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > > wrote: > >> > >> On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza > >> wrote: > >> >> version. First i enter command at my freeswitch box under > >> >> mod_skypopen/configs > >> >> > >> >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 > >> >> > >> >> then I install xauth on server after that i ssh with x forwarding to > my > >> >> server from an another linux desktop and open skype by typing > >> >> /usr/bin/skype, it launched skype client at Linux desktop but it > didn't > >> >> ask > >> >> for connecting with skypopen api, although it creates ".Skype" > >> >> directory on > >> > >> After launching the skype client from the ssh -X session, you have to > >> launch skypopen_auth from the same ssh -X session (giving the X server > >> as an argument), eg: "./skypopen_auth $DISPLAY" > >> > >> >> my server but when i load mod_skypopen it says could not find any > skype > >> >> instance and when i typed skypopen_auth it also says no skype > instance > >> >> found > >> >> on X 0:0. Can anyone tell me where I have mistaken?? > >> > >> After having given the auth to the skype client, and closing it so it > >> save that auth, you close the ssh -X session, and launch an X server > >> and a skype client in the server, using the script in the configs > >> directory (as explained in the wiki). > >> > >> After having launched that script, you load mod_skypopen. > >> > >> You MUST edit the script and the skypopen.conf.xml to use your own > >> values for skype username and password. > >> > >> -- > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Regards, > > Zuhair Raza > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/7fa59d54/attachment-0001.html From david.ponzone at gmail.com Mon Jun 7 04:29:24 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 7 Jun 2010 13:29:24 +0200 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> References: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> Message-ID: Mike, You're right, it can be achieved with SIP now that I think a bit more about it. The idea was to allow adding multiple media gateways when required, so the media gateways should not be facing the carriers as some of them do SIP-filtering, but should only be advertised in the SDP. So SIP-only boxes (doing bypass-media) should face the carriers to handle the trunking. In the middle, we would then have the media gateways, doing SIP and mostly RTP. But I guess we dont want customers to register and to send calls to a media gateway, so we need another set of SIP boxes on the other side, doing bypass-media also. So it would like this: ------sip-----FS-RTP-1-----sip------ FS-SIP-Internal-1 ------sip-----FS-RTP-2-----sip------FS-SIP- External-1----sip-----Carriers ------sip-----FS-RTP-3-----sip------ FS-SIP-Internal-2 -------sip----FS-RTP-4-----sip------FS-SIP- External-2-----sip----Carriers -------sip----FS-RTP-5-----sip------ Thanks to bypass-media, the RTP streams would go from customer to FS- RTP-x to Carriers, and reverse. And I don't see any reason why the same set of FS-SIP boxes could not be used for both internal and external borders. Is there something wrong in this ? Code, does it help ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 05/06/2010 ? 19:54, Michael Jerris a ?crit : > Why would it be an advantage to have your media proxies use another > protocol? > > Mike > > On Jun 4, 2010, at 1:59 AM, David Ponzone wrote: > >> It doesn't solve the issue that all the media servers will do >> signaling too, and will talk SIP with the carriers. >> So the carriers will need to allow all the media servers . >> >> The only clean solution to avoid that, I think, is to have >> signaling boxes allocating resources from media servers with >> another protocol than SIP. >> RTPproxy does that I think, but I am not sure how it works. >> >> David Ponzone > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/1d4a0e66/attachment.html From david.ponzone at gmail.com Mon Jun 7 04:29:42 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 7 Jun 2010 13:29:42 +0200 Subject: [Freeswitch-users] RTP Proxy In-Reply-To: <1275909314.2807.0.camel@gustavo-laptop> References: <1275909314.2807.0.camel@gustavo-laptop> Message-ID: <639452D0-B299-4214-8227-7CCBFAA8D49A@gmail.com> There is one included :) David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 07/06/2010 ? 13:15, Gustavo Espeche a ?crit : > Hi some one know an rtp proxy that work with freeswitch. > Thanks. > > Gustavo Espeche > www.easyipcall.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/0bfe17f9/attachment.html From gmaruzz at celliax.org Mon Jun 7 04:36:34 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 7 Jun 2010 13:36:34 +0200 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: On Mon, Jun 7, 2010 at 1:29 PM, Zuhair Raza wrote: > ok thanks.. but what about my question that skype didnot ask me to to > connect to skype API.? and when i do either on ./skypopen_auth :101 or :0 it > says cannot open X display I told you in previous mail. You need to launch skypopen_auth from the same ssh -X from which you launched the Skype client. And you must give skypopen_auth the correct xserver as an argument. You can use $DISPLAY, or you can check it with "echo $DISPLAY" and then use that value. > > On Mon, Jun 7, 2010 at 3:53 PM, Giovanni Maruzzelli > wrote: >> >> On Mon, Jun 7, 2010 at 12:45 PM, Zuhair Raza >> wrote: >> > Thanks for explanation Sir, One more question, please explain >> > >> > cd /root >> > mount /dev/hda5 /mnt >> > cp /mnt/root/skypeconfig2.tgz ./ >> > tar xzf skypeconfig2.tgz >> > chown -R root.root .Skype >> > >> > According to wiki we haven't created a tgz file before that, but .Skype >> > directory at the server >> >> If you do it with ssh -X (as pre the previous mail), you don't need a >> Skype config directory from another computer. You created that >> directory. >> So, just skip those steps. >> >> -giovanni >> >> > >> > >> > On Mon, Jun 7, 2010 at 12:07 PM, Giovanni Maruzzelli >> > >> > wrote: >> >> >> >> On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza >> >> wrote: >> >> >> version. First i enter command at my freeswitch box under >> >> >> mod_skypopen/configs >> >> >> >> >> >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 >> >> >> >> >> >> then I install xauth on server after that i ssh with x forwarding to >> >> >> my >> >> >> server from an another linux desktop and open skype by typing >> >> >> /usr/bin/skype, it launched skype client at Linux desktop but it >> >> >> didn't >> >> >> ask >> >> >> for connecting with skypopen api, although it creates ".Skype" >> >> >> directory on >> >> >> >> After launching the skype client from the ssh -X session, you have to >> >> launch skypopen_auth from the same ssh -X session (giving the X server >> >> as an argument), eg: "./skypopen_auth $DISPLAY" >> >> >> >> >> my server but when i load mod_skypopen it says could not find any >> >> >> skype >> >> >> instance and when i typed skypopen_auth it also says no skype >> >> >> instance >> >> >> found >> >> >> on X 0:0. Can anyone tell me where I have mistaken?? >> >> >> >> After having given the auth to the skype client, and closing it so it >> >> save that auth, you close the ssh -X session, and launch an X server >> >> and a skype client in the server, using the script in the configs >> >> directory (as explained in the wiki). >> >> >> >> After having launched that script, you load mod_skypopen. >> >> >> >> You MUST edit the script and the skypopen.conf.xml to use your own >> >> values for skype username and password. >> >> >> >> -- >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> Cell : +39-347-2665618 >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Regards, >> > Zuhair Raza >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Regards, > Zuhair Raza > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From thangappan143 at gmail.com Mon Jun 7 04:48:35 2010 From: thangappan143 at gmail.com (Thangappan.M) Date: Mon, 7 Jun 2010 17:18:35 +0530 Subject: [Freeswitch-users] Need to stop more than one voice file using break application In-Reply-To: References: Message-ID: For normal scenario it is working fine. I have done the break application in the signal handler at that time it is not working. Any reason? On Fri, Jun 4, 2010 at 9:03 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I added a patch [eba05c3 ] so it will work now > > > On Fri, Jun 4, 2010 at 4:01 AM, Thangappan.M wrote: > >> >> Dear all, >> >> I am in the process of developing IVR using FreeSWITCH. For that I am >> being used outbound ESL in async mode. >> >> In my design, usually for one menu it might be more than one voice >> files. So using playback_delimiter, play back all the voice file in single >> instance using playback application. >> >> I've tried to stop the play back using "break all" API. But it >> only break the only one voice file not the whole application(playback). >> >> Consider that I am playback four voice files. When the first voice file >> is getting playback, using " break all" t stop the playback . It >> only stopped the second voice and continued to playback the third voice file >> and followed by fourth one. >> >> My need is to break the playback application which may have any number >> of voice files in async mode. >> >> Thanks in advance. >> >> >> -- >> Regards, >> Thangappan.M >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/cca608ae/attachment.html From jan.berger at video24.no Mon Jun 7 05:01:06 2010 From: jan.berger at video24.no (Jan Berger) Date: Mon, 7 Jun 2010 14:01:06 +0200 Subject: [Freeswitch-users] speech recognition? In-Reply-To: References: Message-ID: <36AC79320BB64F82B4E2A91ACA96B054@dell9400> Myron, If you insist on using Simon I will suggest that you get some help from the Simon community to connect mrcp using unimrcp or similar onhttp://wiki.freeswitch.org/wiki/Mod_unimrcp. Once this is available it should just work. Simon is GPL - meaning the license is not compatible with FS - but, that does not prevent you as an end-user of using it. Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Myron Curtis Sent: 5. juni 2010 22:20 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] speech recognition? Hi, I am developing a virtual world called Virtual Worlds Grid, and I am using FreeSwitch to provide voice. I would like to integrate it with a speech recognition engine like Simon, in order to have everything said in voice to be typed into text. This is one way to make the grid accessible for people who cannot use voice, and it also allows language translation software like Open Bablefish, to translate voice conversations from the resulting text files, which would facilitate multilingual conversations. I am new to the FreeSwitch system, and need some guidance on how to set this up. I would appreciate any tips, cautions, or links to practical help I can get. If this will work in a virtual world, there is no reason it cannot be made to work on cell phones. Thanks, Myron Curtis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/ee562ec2/attachment.html From engineerzuhairraza at gmail.com Mon Jun 7 05:05:02 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Mon, 7 Jun 2010 17:05:02 +0500 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: you mean to use this syntax gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:101 ?? When i do echo $DISPLAY it says localhost:10.0 and when gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:$DISPLAY it says cannot find -lX11:localhost:10.0 On Mon, Jun 7, 2010 at 4:36 PM, Giovanni Maruzzelli wrote: > On Mon, Jun 7, 2010 at 1:29 PM, Zuhair Raza > wrote: > > ok thanks.. but what about my question that skype didnot ask me to to > > connect to skype API. and when i do either on ./skypopen_auth :101 or :0 > it > > says cannot open X display > > I told you in previous mail. > > You need to launch skypopen_auth from the same ssh -X from which you > launched the Skype client. And you must give skypopen_auth the correct > xserver as an argument. > You can use $DISPLAY, or you can check it with "echo $DISPLAY" and > then use that value. > > > > > > On Mon, Jun 7, 2010 at 3:53 PM, Giovanni Maruzzelli > > > wrote: > >> > >> On Mon, Jun 7, 2010 at 12:45 PM, Zuhair Raza > >> wrote: > >> > Thanks for explanation Sir, One more question, please explain > >> > > >> > cd /root > >> > mount /dev/hda5 /mnt > >> > cp /mnt/root/skypeconfig2.tgz ./ > >> > tar xzf skypeconfig2.tgz > >> > chown -R root.root .Skype > >> > > >> > According to wiki we haven't created a tgz file before that, but > .Skype > >> > directory at the server > >> > >> If you do it with ssh -X (as pre the previous mail), you don't need a > >> Skype config directory from another computer. You created that > >> directory. > >> So, just skip those steps. > >> > >> -giovanni > >> > >> > > >> > > >> > On Mon, Jun 7, 2010 at 12:07 PM, Giovanni Maruzzelli > >> > > >> > wrote: > >> >> > >> >> On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza > >> >> wrote: > >> >> >> version. First i enter command at my freeswitch box under > >> >> >> mod_skypopen/configs > >> >> >> > >> >> >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 > >> >> >> > >> >> >> then I install xauth on server after that i ssh with x forwarding > to > >> >> >> my > >> >> >> server from an another linux desktop and open skype by typing > >> >> >> /usr/bin/skype, it launched skype client at Linux desktop but it > >> >> >> didn't > >> >> >> ask > >> >> >> for connecting with skypopen api, although it creates ".Skype" > >> >> >> directory on > >> >> > >> >> After launching the skype client from the ssh -X session, you have to > >> >> launch skypopen_auth from the same ssh -X session (giving the X > server > >> >> as an argument), eg: "./skypopen_auth $DISPLAY" > >> >> > >> >> >> my server but when i load mod_skypopen it says could not find any > >> >> >> skype > >> >> >> instance and when i typed skypopen_auth it also says no skype > >> >> >> instance > >> >> >> found > >> >> >> on X 0:0. Can anyone tell me where I have mistaken?? > >> >> > >> >> After having given the auth to the skype client, and closing it so it > >> >> save that auth, you close the ssh -X session, and launch an X server > >> >> and a skype client in the server, using the script in the configs > >> >> directory (as explained in the wiki). > >> >> > >> >> After having launched that script, you load mod_skypopen. > >> >> > >> >> You MUST edit the script and the skypopen.conf.xml to use your own > >> >> values for skype username and password. > >> >> > >> >> -- > >> >> Sincerely, > >> >> > >> >> Giovanni Maruzzelli > >> >> Cell : +39-347-2665618 > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > > >> > -- > >> > Regards, > >> > Zuhair Raza > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Regards, > > Zuhair Raza > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/459bc014/attachment-0001.html From gmaruzz at celliax.org Mon Jun 7 05:19:35 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 7 Jun 2010 14:19:35 +0200 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: Zuhair, maybe you need to find a friend with some more unix experience, that can help you for the first few days. You have to: 1) compile skypopen_auth as in the wiki ("gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11") 2) login with ssh -X 3) launch Skype client 4) launch "skypopen_auth localhost:10.0" -giovanni On Mon, Jun 7, 2010 at 2:05 PM, Zuhair Raza wrote: > you mean to use this syntax > gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:101? ?? > > When i do echo $DISPLAY it says localhost:10.0 > and when gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:$DISPLAY > it says cannot find -lX11:localhost:10.0 > > > On Mon, Jun 7, 2010 at 4:36 PM, Giovanni Maruzzelli > wrote: >> >> On Mon, Jun 7, 2010 at 1:29 PM, Zuhair Raza >> wrote: >> > ok thanks.. but what about my question that skype didnot ask me to to >> > connect to skype API.? and when i do either on ./skypopen_auth :101 or >> > :0 it >> > says cannot open X display >> >> I told you in previous mail. >> >> You need to launch skypopen_auth from the same ssh -X from which you >> launched the Skype client. And you must give skypopen_auth the correct >> xserver as an argument. >> You can use $DISPLAY, or you can check it with "echo $DISPLAY" and >> then use that value. >> >> >> > >> > On Mon, Jun 7, 2010 at 3:53 PM, Giovanni Maruzzelli >> > >> > wrote: >> >> >> >> On Mon, Jun 7, 2010 at 12:45 PM, Zuhair Raza >> >> wrote: >> >> > Thanks for explanation Sir, One more question, please explain >> >> > >> >> > cd /root >> >> > mount /dev/hda5 /mnt >> >> > cp /mnt/root/skypeconfig2.tgz ./ >> >> > tar xzf skypeconfig2.tgz >> >> > chown -R root.root .Skype >> >> > >> >> > According to wiki we haven't created a tgz file before that, but >> >> > .Skype >> >> > directory at the server >> >> >> >> If you do it with ssh -X (as pre the previous mail), you don't need a >> >> Skype config directory from another computer. You created that >> >> directory. >> >> So, just skip those steps. >> >> >> >> -giovanni >> >> >> >> > >> >> > >> >> > On Mon, Jun 7, 2010 at 12:07 PM, Giovanni Maruzzelli >> >> > >> >> > wrote: >> >> >> >> >> >> On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza >> >> >> wrote: >> >> >> >> version. First i enter command at my freeswitch box under >> >> >> >> mod_skypopen/configs >> >> >> >> >> >> >> >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 >> >> >> >> >> >> >> >> then I install xauth on server after that i ssh with x forwarding >> >> >> >> to >> >> >> >> my >> >> >> >> server from an another linux desktop and open skype by typing >> >> >> >> /usr/bin/skype, it launched skype client at Linux desktop but it >> >> >> >> didn't >> >> >> >> ask >> >> >> >> for connecting with skypopen api, although it creates ".Skype" >> >> >> >> directory on >> >> >> >> >> >> After launching the skype client from the ssh -X session, you have >> >> >> to >> >> >> launch skypopen_auth from the same ssh -X session (giving the X >> >> >> server >> >> >> as an argument), eg: "./skypopen_auth $DISPLAY" >> >> >> >> >> >> >> my server but when i load mod_skypopen it says could not find any >> >> >> >> skype >> >> >> >> instance and when i typed skypopen_auth it also says no skype >> >> >> >> instance >> >> >> >> found >> >> >> >> on X 0:0. Can anyone tell me where I have mistaken?? >> >> >> >> >> >> After having given the auth to the skype client, and closing it so >> >> >> it >> >> >> save that auth, you close the ssh -X session, and launch an X server >> >> >> and a skype client in the server, using the script in the configs >> >> >> directory (as explained in the wiki). >> >> >> >> >> >> After having launched that script, you load mod_skypopen. >> >> >> >> >> >> You MUST edit the script and the skypopen.conf.xml to use your own >> >> >> values for skype username and password. >> >> >> >> >> >> -- >> >> >> Sincerely, >> >> >> >> >> >> Giovanni Maruzzelli >> >> >> Cell : +39-347-2665618 >> >> >> >> >> >> _______________________________________________ >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > -- >> >> > Regards, >> >> > Zuhair Raza >> >> > >> >> > >> >> > _______________________________________________ >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at 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 >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Regards, >> > Zuhair Raza >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Regards, > Zuhair Raza > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From darkeagle6 at hotmail.com Mon Jun 7 05:32:23 2010 From: darkeagle6 at hotmail.com (Myron Curtis) Date: Mon, 7 Jun 2010 05:32:23 -0700 Subject: [Freeswitch-users] speech recognition? In-Reply-To: <36AC79320BB64F82B4E2A91ACA96B054@dell9400> References: <36AC79320BB64F82B4E2A91ACA96B054@dell9400> Message-ID: Thank you Jan, I t sounds like you might like a different speech recognition system for FreeSwitch better than Simon. Do you have suggestions for some that might be more compatible? I chose Simon only because it was recommended to me by a friend who has worked with it and he was willing to help set it up. Thanks for your quick reply. Myron From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger Sent: Monday, June 07, 2010 5:01 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] speech recognition? Myron, If you insist on using Simon I will suggest that you get some help from the Simon community to connect mrcp using unimrcp or similar onhttp://wiki.freeswitch.org/wiki/Mod_unimrcp. Once this is available it should just work. Simon is GPL - meaning the license is not compatible with FS - but, that does not prevent you as an end-user of using it. Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Myron Curtis Sent: 5. juni 2010 22:20 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] speech recognition? Hi, I am developing a virtual world called Virtual Worlds Grid, and I am using FreeSwitch to provide voice. I would like to integrate it with a speech recognition engine like Simon, in order to have everything said in voice to be typed into text. This is one way to make the grid accessible for people who cannot use voice, and it also allows language translation software like Open Bablefish, to translate voice conversations from the resulting text files, which would facilitate multilingual conversations. I am new to the FreeSwitch system, and need some guidance on how to set this up. I would appreciate any tips, cautions, or links to practical help I can get. If this will work in a virtual world, there is no reason it cannot be made to work on cell phones. Thanks, Myron Curtis No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.829 / Virus Database: 271.1.1/2923 - Release Date: 06/06/10 23:35:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/8b770e05/attachment.html From darkeagle6 at hotmail.com Mon Jun 7 05:34:40 2010 From: darkeagle6 at hotmail.com (Myron Curtis) Date: Mon, 7 Jun 2010 05:34:40 -0700 Subject: [Freeswitch-users] speech recognition? In-Reply-To: <7588B1B7A1B741A4A2A72C38249FC92D@dell9400> References: <7588B1B7A1B741A4A2A72C38249FC92D@dell9400> Message-ID: I will look more closely at the wiki to see if pcketsphinx + bridging will do what I need. Thanks, Myron From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger Sent: Monday, June 07, 2010 3:46 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] speech recognition? Hi It was someone else who requested a similar natural speech to text service earlier.... It is some ASR examples on wiki - pocketsphinx exist + bridging into commercial versions. What is "Simon" ? Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Myron Curtis Sent: 5. juni 2010 22:20 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] speech recognition? Hi, I am developing a virtual world called Virtual Worlds Grid, and I am using FreeSwitch to provide voice. I would like to integrate it with a speech recognition engine like Simon, in order to have everything said in voice to be typed into text. This is one way to make the grid accessible for people who cannot use voice, and it also allows language translation software like Open Bablefish, to translate voice conversations from the resulting text files, which would facilitate multilingual conversations. I am new to the FreeSwitch system, and need some guidance on how to set this up. I would appreciate any tips, cautions, or links to practical help I can get. If this will work in a virtual world, there is no reason it cannot be made to work on cell phones. Thanks, Myron Curtis No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.829 / Virus Database: 271.1.1/2923 - Release Date: 06/06/10 23:35:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/0d894b5b/attachment-0001.html From oliver.schenk at iinet.net.au Mon Jun 7 06:21:06 2010 From: oliver.schenk at iinet.net.au (Oliver Schenk) Date: Mon, 07 Jun 2010 21:21:06 +0800 Subject: [Freeswitch-users] Freeswitch kills my home voip phone Message-ID: <4C0CF242.40301@iinet.net.au> Hi All, I'm a total freeswitch noob at the moment. I was playing at home with FreeSwitch on Windows compiled from source. I have a router at home that registers with my ISP's voip SIP server. My hard phone is connected to that router to make calls. Whenever I have FreeSwitch running on my PC (which is on the same LAN as the router) I lose my VOIP connection to the ISP and I can't re-register until I reboot my internet connection. This happens straight out of the box. Why is this happening? Also while freeswitch is running I can't seem to make any calls ... is Freeswitch hijacking my connection? Thanks, Oliver From jan.berger at video24.no Mon Jun 7 06:54:56 2010 From: jan.berger at video24.no (Jan Berger) Date: Mon, 7 Jun 2010 15:54:56 +0200 Subject: [Freeswitch-users] speech recognition? In-Reply-To: References: <36AC79320BB64F82B4E2A91ACA96B054@dell9400> Message-ID: I don't have any preferences. Anything that works goes - but I like the idea of a standard plug-in interface on this area. Most countries have speech detection, but you will not find a single vendor that supports all languages. In your case I would look for a natural speech engine - something I am not to familiar with myself. My last attempt on doing that gave some very interesting results :-) Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Myron Curtis Sent: 7. juni 2010 14:32 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] speech recognition? Thank you Jan, I t sounds like you might like a different speech recognition system for FreeSwitch better than Simon. Do you have suggestions for some that might be more compatible? I chose Simon only because it was recommended to me by a friend who has worked with it and he was willing to help set it up. Thanks for your quick reply. Myron From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger Sent: Monday, June 07, 2010 5:01 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] speech recognition? Myron, If you insist on using Simon I will suggest that you get some help from the Simon community to connect mrcp using unimrcp or similar onhttp://wiki.freeswitch.org/wiki/Mod_unimrcp. Once this is available it should just work. Simon is GPL - meaning the license is not compatible with FS - but, that does not prevent you as an end-user of using it. Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Myron Curtis Sent: 5. juni 2010 22:20 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] speech recognition? Hi, I am developing a virtual world called Virtual Worlds Grid, and I am using FreeSwitch to provide voice. I would like to integrate it with a speech recognition engine like Simon, in order to have everything said in voice to be typed into text. This is one way to make the grid accessible for people who cannot use voice, and it also allows language translation software like Open Bablefish, to translate voice conversations from the resulting text files, which would facilitate multilingual conversations. I am new to the FreeSwitch system, and need some guidance on how to set this up. I would appreciate any tips, cautions, or links to practical help I can get. If this will work in a virtual world, there is no reason it cannot be made to work on cell phones. Thanks, Myron Curtis No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.829 / Virus Database: 271.1.1/2923 - Release Date: 06/06/10 23:35:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/1ce1d402/attachment.html From rupa at rupa.com Mon Jun 7 06:56:03 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 7 Jun 2010 08:56:03 -0500 Subject: [Freeswitch-users] Freeswitch kills my home voip phone In-Reply-To: <4C0CF242.40301@iinet.net.au> References: <4C0CF242.40301@iinet.net.au> Message-ID: Try starting with -nonat to see if this solves your problem. My guess is that FS is setting up your router to forward anything for port 5060 to itself which causes your hard phone to no longer get external connections. If that works, change the port for the internal profile to a port other than 5060. On Mon, Jun 7, 2010 at 8:21 AM, Oliver Schenk wrote: > Hi All, > > I'm a total freeswitch noob at the moment. > > I was playing at home with FreeSwitch on Windows compiled from source. I > have a router at home that registers with my ISP's voip SIP server. My > hard phone is connected to that router to make calls. > > Whenever I have FreeSwitch running on my PC (which is on the same LAN as > the router) I lose my VOIP connection to the ISP and I can't re-register > until I reboot my internet connection. This happens straight out of the > box. Why is this happening? Also while freeswitch is running I can't > seem to make any calls ... is Freeswitch hijacking my connection? > > Thanks, > > Oliver > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/f469ce4b/attachment.html From asannucci at gmail.com Mon Jun 7 06:59:24 2010 From: asannucci at gmail.com (bakko) Date: Mon, 7 Jun 2010 15:59:24 +0200 Subject: [Freeswitch-users] Skypopen problem In-Reply-To: References: Message-ID: <8EE5FE8AA51748EEA666BC23331B2B3E@voztovoice> Ciao Giovanni :) after the reboot the problem still lives :) my dmesg: snd: version magic '2.6.18.8-linode22 SMP mod_unload Xen PENTIUM4 REGPARM gcc-4.1' should be '2.6.18.8-linode22 SMP mod_unload Xen PENTIUM4 REGPARM gcc-4.3' snd: no version magic, tainting kernel. snd: Unknown symbol unregister_sound_special snd: Unknown symbol register_sound_special_device snd: Unknown symbol sound_class snd: no version magic, tainting kernel. snd: Unknown symbol unregister_sound_special snd: Unknown symbol register_sound_special_device snd: Unknown symbol sound_class snd: no version magic, tainting kernel. snd: Unknown symbol unregister_sound_special snd: Unknown symbol register_sound_special_device snd: Unknown symbol sound_class Maybe can be the kernel but i don?t know how resolve. Grazie mille per l'aiuto. Ciao - Andrea - From gmaruzz at celliax.org Mon Jun 7 07:21:59 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 7 Jun 2010 16:21:59 +0200 Subject: [Freeswitch-users] Skypopen problem In-Reply-To: <8EE5FE8AA51748EEA666BC23331B2B3E@voztovoice> References: <8EE5FE8AA51748EEA666BC23331B2B3E@voztovoice> Message-ID: On Mon, Jun 7, 2010 at 3:59 PM, bakko wrote: > after the reboot the problem still lives :) > > my dmesg: > > snd: version magic '2.6.18.8-linode22 SMP mod_unload Xen PENTIUM4 REGPARM > gcc-4.1' should be '2.6.18.8-linode22 SMP mod_unload Xen PENTIUM4 REGPARM > gcc-4.3' > snd: no version magic, tainting kernel. > snd: Unknown symbol unregister_sound_special > snd: Unknown symbol register_sound_special_device > snd: Unknown symbol sound_class > snd: no version magic, tainting kernel. > snd: Unknown symbol unregister_sound_special > snd: Unknown symbol register_sound_special_device > snd: Unknown symbol sound_class > snd: no version magic, tainting kernel. > snd: Unknown symbol unregister_sound_special > snd: Unknown symbol register_sound_special_device > snd: Unknown symbol sound_class Ciao Andrea, maybe is because you're not linking with the correct header files for your kernel? Are you sure you installed the kernel-headers of the kernel you got from linode? Also, if you have the correct kernel headers, could you ask in the linode forum how is the correct procedure to compile alsa (or any other module) for that kernel, and report here? (so I can add it to the wiki page). Sorry of not been of more help, a presto, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From asannucci at gmail.com Mon Jun 7 07:44:54 2010 From: asannucci at gmail.com (bakko) Date: Mon, 7 Jun 2010 16:44:54 +0200 Subject: [Freeswitch-users] Skypopen problem In-Reply-To: References: <8EE5FE8AA51748EEA666BC23331B2B3E@voztovoice> Message-ID: <762FE0D09FCF40B69D16E0AC1064AAC0@voztovoice> Than you Giovanni. I hope to have some news very soon and learn inglish more fast :) BR - Andrea - From brian at freeswitch.org Mon Jun 7 07:58:22 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Jun 2010 09:58:22 -0500 Subject: [Freeswitch-users] Change in behaviour for ESL events? In-Reply-To: <1275896373992-5147891.post@n2.nabble.com> References: <1275896373992-5147891.post@n2.nabble.com> Message-ID: You'll need to pull the source from git as this was fixed late last night. /b On Jun 7, 2010, at 2:39 AM, peely wrote: > > Hi, > > In the latest git snapshot I've compiled, event subscriptions within ESL no > longer seem to function in the same way. > > I used to issue "filter Unique-ID {uuid}\n\n" and "filter > Other-Leg-Unique-ID {uuid}\n\n" followed by "events plain all\n\n". I did > this because "myevents\n\n" would not allow me to subscribe to events for > background jobs issued by bgapi, which I do quite a lot. Applying the filter > then subscribing to all events seemed the most stable and allowed me to > subscribe to additional events should I need them. > > In the latest snapshot, I don't receive any events through this mechanism. I > tried "event text all" as newly suggested on the ESL outbound wiki page, but > this transmits a heap of white space to my socket then kills freeswitch! > > Could somebody please tell me if this is something that is "work in > progress" and will ultimately resume old behaviour, or should I be doing > something else to monitor events for my uuid and anything I spawn in that > session? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/f7f16022/attachment.html From anthony.minessale at gmail.com Mon Jun 7 08:01:06 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Jun 2010 10:01:06 -0500 Subject: [Freeswitch-users] speech recognition? In-Reply-To: References: <36AC79320BB64F82B4E2A91ACA96B054@dell9400> Message-ID: you would need one that was designed for dictation, pocketsphinx only works on small dictionaries that were chosen for the application at hand. On Mon, Jun 7, 2010 at 8:54 AM, Jan Berger wrote: > I don?t have any preferences. Anything that works goes ? but I like the > idea of a standard plug-in interface on this area. Most countries have > speech detection, but you will not find a single vendor that supports all > languages. > > > > In your case I would look for a natural speech engine ? something I am not > to familiar with myself. My last attempt on doing that gave some very > interesting results J > > > > Jan > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Myron Curtis > *Sent:* 7. juni 2010 14:32 > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] speech recognition? > > > > Thank you Jan, > > I t sounds like you might like a different speech recognition system for > FreeSwitch better than Simon. Do you have suggestions for some that might be > more compatible? I chose Simon only because it was recommended to me by a > friend who has worked with it and he was willing to help set it up. > > Thanks for your quick reply. > > Myron > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jan Berger > *Sent:* Monday, June 07, 2010 5:01 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] speech recognition? > > > > Myron, > > > > If you insist on using Simon I will suggest that you get some help from the > Simon community to connect mrcp using unimrcp or similar onhttp:// > wiki.freeswitch.org/wiki/Mod_unimrcp. Once this is available it should > just work. > > > > Simon is GPL ? meaning the license is not compatible with FS ? but, that > does not prevent you as an end-user of using it. > > > > Jan > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Myron Curtis > *Sent:* 5. juni 2010 22:20 > *To:* FreeSWITCH-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] speech recognition? > > > > > > > > Hi, > > I am developing a virtual world called Virtual Worlds Grid, and I am using > FreeSwitch to provide voice. I would like to integrate it with a speech > recognition engine like Simon, in order to have everything said in voice to > be typed into text. This is one way to make the grid accessible for people > who cannot use voice, and it also allows language translation software like > Open Bablefish, to translate voice conversations from the resulting text > files, which would facilitate multilingual conversations. > > I am new to the FreeSwitch system, and need some guidance on how to set > this up. > > I would appreciate any tips, cautions, or links to practical help I can > get. > > If this will work in a virtual world, there is no reason it cannot be made > to work on cell phones. > > Thanks, > > Myron Curtis > > > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 9.0.829 / Virus Database: 271.1.1/2923 - Release Date: 06/06/10 > 23:35:00 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100607/4dbc045a/attachment-0001.html From gustavo.espeche at upper-soft.com Mon Jun 7 08:36:35 2010 From: gustavo.espeche at upper-soft.com (Gustavo Espeche) Date: Mon, 07 Jun 2010 12:36:35 -0300 Subject: [Freeswitch-users] RTP Proxy In-Reply-To: <639452D0-B299-4214-8227-7CCBFAA8D49A@gmail.com> References: <1275909314.2807.0.camel@gustavo-laptop> <639452D0-B299-4214-8227-7CCBFAA8D49A@gmail.com> Message-ID: <1275924995.2807.9.camel@gustavo-laptop> OK i know it, but i'm trying to configure an rtp-proxy in a different server that is running freeswitch. Gustavo Espeche www.easyipcall.com On Mon, 2010-06-07 at 13:29 +0200, David Ponzone wrote: > There is one included :) > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > ? l'intention exclusive de ses destinataires. Toute utilisation ou > diffusion non autoris?e est interdite. Tout message ?lectronique est > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes > pas destinataire de ce message, merci de le d?truire imm?diatement et > d'avertir l'exp?diteur. > > > > > > > > Le 07/06/2010 ? 13:15, Gustavo Espeche a ?crit : > > > Hi some one know an rtp proxy that work with freeswitch. > > Thanks. > > > > Gustavo Espeche > > www.easyipcall.com > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 david.ponzone at gmail.com Mon Jun 7 08:47:39 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 7 Jun 2010 17:47:39 +0200 Subject: [Freeswitch-users] RTP Proxy In-Reply-To: <1275924995.2807.9.camel@gustavo-laptop> References: <1275909314.2807.0.camel@gustavo-laptop> <639452D0-B299-4214-8227-7CCBFAA8D49A@gmail.com> <1275924995.2807.9.camel@gustavo-laptop> Message-ID: Perhaps can you explain exactly what you're trying to achieve. It would help the people here to help you :) David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 07/06/2010 ? 17:36, Gustavo Espeche a ?crit : > OK i know it, but i'm trying to configure an rtp-proxy in a different > server that is running freeswitch. > > Gustavo Espeche > www.easyipcall.com > > On Mon, 2010-06-07 at 13:29 +0200, David Ponzone wrote: >> There is one included :) >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis >> ? l'intention exclusive de ses destinataires. Toute utilisation ou >> diffusion non autoris?e est interdite. Tout message ?lectronique >> est >> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au >> titre >> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous >> n'?tes >> pas destinataire de ce message, merci de le d?truire imm?diatement >> et >> d'avertir l'exp?diteur. >> >> >> >> >> >> >> >> Le 07/06/2010 ? 13:15, Gustavo Espeche a ?crit : >> >>> Hi some one know an rtp proxy that work with freeswitch. >>> Thanks. >>> >>> Gustavo Espeche >>> www.easyipcall.com >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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/20100607/4c81437f/attachment.html From msc at freeswitch.org Mon Jun 7 09:37:08 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 09:37:08 -0700 Subject: [Freeswitch-users] outbound socket problem / fs_ivrd general protection, segfault In-Reply-To: <4C092510.7040202@xpirio.com> References: <4C08C3E2.3020703@xpirio.com> <4C091026.5040209@xpirio.com> <7F6151C9-4819-471E-B962-552DDB324253@freeswitch.org> <4C091619.9020200@xpirio.com> <4C092510.7040202@xpirio.com> Message-ID: 2010/6/4 Christian L?schenkohl > thank you very much > > updating to the latest version did cure all my problems > i will stay on trunk from now on > > sorry that i didn't try this as first step > > br > > FYI, I use this shell script on a few test boxes. I keep it in the root of the src tree and run it whenever I want to update to the latest, which is usually each morning when I arrive at my desk. <9>:cat rebuild-freeswitch.sh #!/bin/sh freeswitch -stop make current cd libs/esl make make perlmod cd ../.. sleep 10 freeswitch -nc -nonat Be sure to build all the ESL-ish mods that you use. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/1b58126b/attachment.html From gmaruzz at celliax.org Mon Jun 7 09:44:01 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 7 Jun 2010 18:44:01 +0200 Subject: [Freeswitch-users] Skypopen problem In-Reply-To: <762FE0D09FCF40B69D16E0AC1064AAC0@voztovoice> References: <8EE5FE8AA51748EEA666BC23331B2B3E@voztovoice> <762FE0D09FCF40B69D16E0AC1064AAC0@voztovoice> Message-ID: On Mon, Jun 7, 2010 at 4:44 PM, bakko wrote: > I hope to have some news very soon and learn inglish more fast :) > we're all using Basic English (600 words), barred a minority that knows Real English (and nobody understand them ;) ). Ciao e a presto! -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From Prometheus001 at gmx.net Mon Jun 7 09:56:16 2010 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 07 Jun 2010 18:56:16 +0200 Subject: [Freeswitch-users] Sending NOTIFY message to a phone via XML-RPC In-Reply-To: <4C08C995.8060500@gmx.net> References: <4C08C995.8060500@gmx.net> Message-ID: <4C0D24B0.3010708@gmx.net> Nobody has a working XML sample for sending NOTIFY events? The wiki is quite lean on this. I will update the wiki with a working sample as soon as I have one. So any suggestions? Best regards Peter Peter P GMX schrieb: > Hello, > > I try to send a NOTIFY message from my application to a registered phone > via XML-RPC. > This is the XML which is sent: > > > > freeswitch.api > > > > sendevent > > > > > NOTIFY,profile=internal,event-string=check-sync;reboot=false,user=200,host=192.168.178.220,content-type=application/simple-message-summary > > > > > > However I receive an error message: > . > . > . > ERROR!. > . > . > > I also tried to send the data as a struct but FS complains about needing > a string as parameters. > > What am I doing wrong? Anybody has a valid sample XML for a NOTIFY message? > > Best regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Mon Jun 7 10:01:05 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 10:01:05 -0700 Subject: [Freeswitch-users] Freeswicth nms In-Reply-To: References: <1111639352-1275526834-cardhu_decombobulator_blackberry.rim.net-2040530345-@bda2058.bisx.prodap.on.blackberry> Message-ID: On Fri, Jun 4, 2010 at 8:53 AM, Phillip Jones wrote: > Are there any FreeSWITCH specific plugins available for Nagios? I started a skeleton of a Perl-based Nagios plugin. You're welcome to help me push that into the realm of actual usability if you are so inclined. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/057377ef/attachment-0001.html From pjintheusa at gmail.com Mon Jun 7 10:02:16 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 7 Jun 2010 13:02:16 -0400 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: References: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> Message-ID: David, I hope you don't mind me interjecting here. But what is the advantage of your setup over the traditional SIP Proxy - FS - SIP Proxy setup? Isn't introducing a re-invite here, to shift the media from FS-SIP-Internal-1 to FS-RTP-3 introducing a further complication and potential point of failure? Pj On Mon, Jun 7, 2010 at 7:29 AM, David Ponzone wrote: > Mike, > > You're right, it can be achieved with SIP now that I think a bit more about > it. > The idea was to allow adding multiple media gateways when required, so the > media gateways should not be facing the carriers as some of them do > SIP-filtering, but should only be advertised in the SDP. > > So SIP-only boxes (doing bypass-media) should face the carriers to handle > the trunking. > In the middle, we would then have the media gateways, doing SIP and mostly > RTP. > But I guess we dont want customers to register and to send calls to a media > gateway, so we need another set of SIP boxes on the other side, doing > bypass-media also. > > So it would like this: > > ------sip-----FS-RTP-1-----sip------ > FS-SIP-Internal-1 > ------sip-----FS-RTP-2-----sip------FS-SIP-External-1----sip-----Carriers > ------sip-----FS-RTP-3-----sip------ > FS-SIP-Internal-2 > -------sip----FS-RTP-4-----sip------FS-SIP-External-2-----sip----Carriers > -------sip----FS-RTP-5-----sip------ > > Thanks to bypass-media, the RTP streams would go from customer to FS-RTP-x > to Carriers, and reverse. > And I don't see any reason why the same set of FS-SIP boxes could not be > used for both internal and external borders. > > Is there something wrong in this ? > > Code, does it help ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 05/06/2010 ? 19:54, Michael Jerris a ?crit : > > Why would it be an advantage to have your media proxies use another > protocol? > > Mike > > On Jun 4, 2010, at 1:59 AM, David Ponzone wrote: > > It doesn't solve the issue that all the media servers will do signaling > too, and will talk SIP with the carriers. > So the carriers will need to allow all the media servers . > > The only clean solution to avoid that, I think, is to have signaling boxes > allocating resources from media servers with another protocol than SIP. > RTPproxy does that I think, but I am not sure how it works. > > David Ponzone > > _______________________________________________ > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100607/3d3e0573/attachment.html From mr.tom.strickland at gmail.com Mon Jun 7 07:02:22 2010 From: mr.tom.strickland at gmail.com (tom strickland) Date: Mon, 7 Jun 2010 15:02:22 +0100 Subject: [Freeswitch-users] Direct inward dialling In-Reply-To: References: <491B305B.4060307@gmx.net> <072A175C-9004-4C14-90EC-9A93A8453787@gmail.com> <35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org> Message-ID: Perhaps you could use an online tool to test your regexes? If I understand correctly, the regexes in FS are all Perl-style regexes and there are a number of online or downloadable tools that will allow you to see the effect of your developing regex on a sample set of data. If this is the case, could anyone recommend a tool? Tom On 2 June 2010 22:17, RR wrote: > > > On Wed, Jun 2, 2010 at 2:17 PM, Michael Collins wrote: > >> >> >> On Tue, Jun 1, 2010 at 7:57 PM, Ron McLeod wrote: >> >>> Is [32|48|54|55|65] correct? Shouldn?t it be (32|48|54|55|65) >>> instead? >>> >>> >>> >>> >>> >>> ^\+?1?(0[0-1]+)?((32|48|54|55|65)\d+)\;?(phone-context=)?\+?(\d+)?$ >>> >> Also, you really don't need the dash in [0-1], just do [01] which means >> "match a 0 or a 1" >> -MC >> >> > Ok cool. Thanks for that. BTW, I've noticed that the regex CLI command > doesn't always behave the same way as the core FS engine. A lot of tests > that display a different result when testing using the regex command yield > different behaviour when one makes a call and how FS behaves/treats the > call. Maybe it's a bug in the old version. We're using a very old version of > FS > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/b5089c2e/attachment-0001.html From ashley at midletearth.com Mon Jun 7 09:59:28 2010 From: ashley at midletearth.com (Ashley B) Date: Mon, 7 Jun 2010 18:59:28 +0200 Subject: [Freeswitch-users] Change in behaviour for ESL events? In-Reply-To: References: <1275896373992-5147891.post@n2.nabble.com> Message-ID: <012001cb0662$cdbae390$6930aab0$@com> Hi Brian, Would this issue have affected not being to receive any "MESSAGE" events when binding to "all" events using EventConsumer in mod_managed? I receive every event BUT "MESSAGE" (afaik). Thanks Ashley From: Brian West [mailto:brian at freeswitch.org] Sent: 07 June 2010 04:58 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Change in behaviour for ESL events? You'll need to pull the source from git as this was fixed late last night. /b On Jun 7, 2010, at 2:39 AM, peely wrote: Hi, In the latest git snapshot I've compiled, event subscriptions within ESL no longer seem to function in the same way. I used to issue "filter Unique-ID {uuid}\n\n" and "filter Other-Leg-Unique-ID {uuid}\n\n" followed by "events plain all\n\n". I did this because "myevents\n\n" would not allow me to subscribe to events for background jobs issued by bgapi, which I do quite a lot. Applying the filter then subscribing to all events seemed the most stable and allowed me to subscribe to additional events should I need them. In the latest snapshot, I don't receive any events through this mechanism. I tried "event text all" as newly suggested on the ESL outbound wiki page, but this transmits a heap of white space to my socket then kills freeswitch! Could somebody please tell me if this is something that is "work in progress" and will ultimately resume old behaviour, or should I be doing something else to monitor events for my uuid and anything I spawn in that session? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/c8948882/attachment-0001.html From msc at freeswitch.org Mon Jun 7 10:03:15 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 10:03:15 -0700 Subject: [Freeswitch-users] get your job aguinst freeswitch In-Reply-To: References: <4C0BEA21.3000008@gmail.com> Message-ID: On Sat, Jun 5, 2010 at 4:11 PM, Moises Silva wrote: > Wow, very nice story ... > > Telephony also changed my life, from working as an intern in a micro voip > startup in Guadalajara, Mexico writing PHP AGI scripts to work happily now > in Sangoma hacking on open source telephony every day :-) > Of course, you had to to move from the nice warm climate in Guad all the way to the frozen north of Canada. Your sacrifice is truly appreciated. :P -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/3223c173/attachment.html From msc at freeswitch.org Mon Jun 7 10:06:22 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 10:06:22 -0700 Subject: [Freeswitch-users] FS "Holds On" to ITSP's 183 for 2 Sec.Before Relaying to Phone? In-Reply-To: References: Message-ID: On Fri, Jun 4, 2010 at 10:28 AM, Alan Frisch wrote: > Brian/Anthony, > > Looks like I forgot to remove the <-- and --> from around the custom > RTP settings. This may have been causing the hang and calls are now > getting full early media, without cut-offs. > > Thanks and have a great weekend! > > Hehe, welcome to XML. :) You'll be used to it in no time. Just be sure to use a text editor with syntax highlighting. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/38c24c3c/attachment.html From brian at freeswitch.org Mon Jun 7 10:07:24 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Jun 2010 12:07:24 -0500 Subject: [Freeswitch-users] Change in behaviour for ESL events? In-Reply-To: <012001cb0662$cdbae390$6930aab0$@com> References: <1275896373992-5147891.post@n2.nabble.com> <012001cb0662$cdbae390$6930aab0$@com> Message-ID: <3DCB9EEF-B62B-41C5-9B15-593AABC9FDE8@freeswitch.org> chances are yes you wouldn't get a single message.... and possibly segfault. /b On Jun 7, 2010, at 11:59 AM, Ashley B wrote: > Hi Brian, > > Would this issue have affected not being to receive any ?MESSAGE? events when binding to ?all? events using EventConsumer in mod_managed? I receive every event BUT ?MESSAGE? (afaik). > > Thanks > Ashley > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/4175ff33/attachment.html From anthony.minessale at gmail.com Mon Jun 7 10:08:05 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Jun 2010 12:08:05 -0500 Subject: [Freeswitch-users] Need to stop more than one voice file using break application In-Reply-To: References: Message-ID: I didn't understand you. On Mon, Jun 7, 2010 at 6:48 AM, Thangappan.M wrote: > For normal scenario it is working fine. I have done the break application > in the signal handler at that time it is not working. > > Any reason? > > > > > On Fri, Jun 4, 2010 at 9:03 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> I added a patch [eba05c3 ] so it will work now >> >> >> On Fri, Jun 4, 2010 at 4:01 AM, Thangappan.M wrote: >> >>> >>> Dear all, >>> >>> I am in the process of developing IVR using FreeSWITCH. For that I am >>> being used outbound ESL in async mode. >>> >>> In my design, usually for one menu it might be more than one voice >>> files. So using playback_delimiter, play back all the voice file in single >>> instance using playback application. >>> >>> I've tried to stop the play back using "break all" API. But it >>> only break the only one voice file not the whole application(playback). >>> >>> Consider that I am playback four voice files. When the first voice file >>> is getting playback, using " break all" t stop the playback . It >>> only stopped the second voice and continued to playback the third voice file >>> and followed by fourth one. >>> >>> My need is to break the playback application which may have any number >>> of voice files in async mode. >>> >>> Thanks in advance. >>> >>> >>> -- >>> Regards, >>> Thangappan.M >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100607/34ac8210/attachment.html From msc at freeswitch.org Mon Jun 7 10:10:07 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 10:10:07 -0700 Subject: [Freeswitch-users] simple fees bill In-Reply-To: <1915997343-1275875496-cardhu_decombobulator_blackberry.rim.net-298651853-@bda057.bisx.prodap.on.blackberry> References: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> <1915997343-1275875496-cardhu_decombobulator_blackberry.rim.net-298651853-@bda057.bisx.prodap.on.blackberry> Message-ID: On Sun, Jun 6, 2010 at 6:51 PM, Budi wibowo wrote: > Anybody running nibble bill on centos 4? > I try to install it but always failed in supporting libraries. > > FYI, Darren Schreiber is scheduled to talk about mod_nibblebill this coming Wednesday on the FS community conf call: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_09 If you can't attend the call then look for the audio recording on the wiki on Thursday. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/c82f6418/attachment.html From stephen at stephenjc.com Mon Jun 7 10:17:26 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Mon, 7 Jun 2010 13:17:26 -0400 Subject: [Freeswitch-users] Direct inward dialling In-Reply-To: References: <491B305B.4060307@gmx.net> <072A175C-9004-4C14-90EC-9A93A8453787@gmail.com> <35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org> Message-ID: i use this tool http://www.gskinner.com/RegExr/desktop/ Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print On Mon, Jun 7, 2010 at 10:02 AM, tom strickland wrote: > Perhaps you could use an online tool to test your regexes? If I understand > correctly, the regexes in FS are all Perl-style regexes and there are a > number of online or downloadable tools that will allow you to see the effect > of your developing regex on a sample set of data. If this is the case, could > anyone recommend a tool? > > Tom > > On 2 June 2010 22:17, RR wrote: > >> >> >> On Wed, Jun 2, 2010 at 2:17 PM, Michael Collins wrote: >> >>> >>> >>> On Tue, Jun 1, 2010 at 7:57 PM, Ron McLeod >> > wrote: >>> >>>> Is [32|48|54|55|65] correct? Shouldn?t it be (32|48|54|55|65) >>>> instead? >>>> >>>> >>>> >>>> >>>> >>>> ^\+?1?(0[0-1]+)?((32|48|54|55|65)\d+)\;?(phone-context=)?\+?(\d+)?$ >>>> >>> Also, you really don't need the dash in [0-1], just do [01] which means >>> "match a 0 or a 1" >>> -MC >>> >>> >> Ok cool. Thanks for that. BTW, I've noticed that the regex CLI command >> doesn't always behave the same way as the core FS engine. A lot of tests >> that display a different result when testing using the regex command yield >> different behaviour when one makes a call and how FS behaves/treats the >> call. Maybe it's a bug in the old version. We're using a very old version of >> FS >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100607/2f8b4f2d/attachment-0001.html From dswardstrom at remotelink.com Mon Jun 7 10:22:30 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Mon, 7 Jun 2010 12:22:30 -0500 (CDT) Subject: [Freeswitch-users] mod_conference Wiki - lists In-Reply-To: <1589313449.150.1275931207129.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <1548359777.152.1275931350885.JavaMail.root@srvr12.remotelinkml.com> I have been looking at the mod_conference Wiki page and have checked some of it's information against the current code in mod_conference.c. In particular, I have been interested in the ?list? capability which is supported by conference_list(). The following comments will be used to change the ?Mod Conference? Wiki page: This applies to the API reference, list entry: Output: First line Conference ( member ) Following lines are csv separated list for each conference leg with the following items: id of participiant (starts at 1 after FS restart) Register string of participiant Uuid of participiants call leg Cid number Cid name Status (hear|speak|floor) Volume In Volume Out Energy Level Output: First line Conference ( member[s][ locked]) Following lines are csv separated list for each conference leg with the following items: id of participant (starts at 1 after FS restart) Register string of participant Uuid of participants call leg Caller id number Caller id name Status (hear|speak|talking|video|floor) Volume In Volume Out Energy Level Notes: First line: ?locked? reflects the lock/unlock status of the conference. Status: ?hear? reflects the mute/unmute status of the member. Status: ?speak? reflects the ?deaf /undeaf? status of the member. Status: ?talking? indicates that the input channel is providing some amount of sound energy. Status: ?video? ??? Status: ?floor? indicates that this member currently owns the floor Does anyone have any comments I should add for floor? Is there anything significant to owning the floor? How about video? It seems that video and video conferencing is a capability that is desired and some support is provided. But is it really working? Does anyone use it? Is there an assumption in conferencing that video is currently sourced from only one place? Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom From gustavo.espeche at upper-soft.com Mon Jun 7 10:32:36 2010 From: gustavo.espeche at upper-soft.com (Gustavo Espeche) Date: Mon, 07 Jun 2010 14:32:36 -0300 Subject: [Freeswitch-users] RTP Proxy In-Reply-To: References: <1275909314.2807.0.camel@gustavo-laptop> <639452D0-B299-4214-8227-7CCBFAA8D49A@gmail.com> <1275924995.2807.9.camel@gustavo-laptop> Message-ID: <1275931956.2807.26.camel@gustavo-laptop> basically i need split the sip signaling and sip media, one server with FS for sip signaling and other one with proxy rtp. i attached a png. If some one work with FS in this schema, please let me know what rtp-proxy do you use, this information can save to me lot of time testing rtp-proxy. Regards. Gustavo Espeche www.easyipcall.com On Mon, 2010-06-07 at 17:47 +0200, David Ponzone wrote: > Perhaps can you explain exactly what you're trying to achieve. It > would help the people here to help you :) > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > ? l'intention exclusive de ses destinataires. Toute utilisation ou > diffusion non autoris?e est interdite. Tout message ?lectronique est > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes > pas destinataire de ce message, merci de le d?truire imm?diatement et > d'avertir l'exp?diteur. > > > > > > > > Le 07/06/2010 ? 17:36, Gustavo Espeche a ?crit : > > > OK i know it, but i'm trying to configure an rtp-proxy in a > > different > > server that is running freeswitch. > > > > Gustavo Espeche > > www.easyipcall.com > > > > On Mon, 2010-06-07 at 13:29 +0200, David Ponzone wrote: > > > There is one included :) > > > > > > David Ponzone Direction Technique > > > email: david.ponzone at ipeva.fr > > > tel: 01 74 03 18 97 > > > gsm: 06 66 98 76 34 > > > > > > > > > Service Client IPeva > > > tel: 0811 46 26 26 > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > ?tablis > > > ? l'intention exclusive de ses destinataires. Toute utilisation ou > > > diffusion non autoris?e est interdite. Tout message ?lectronique > > > est > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > > titre > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > > n'?tes > > > pas destinataire de ce message, merci de le d?truire imm?diatement > > > et > > > d'avertir l'exp?diteur. > > > > > > > > > > > > > > > > > > > > > > > > Le 07/06/2010 ? 13:15, Gustavo Espeche a ?crit : > > > > > > > Hi some one know an rtp proxy that work with freeswitch. > > > > Thanks. > > > > > > > > Gustavo Espeche > > > > www.easyipcall.com > > > > > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.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 -------------- A non-text attachment was scrubbed... Name: FS_proxy_media.png Type: image/png Size: 24375 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/e7deb293/attachment-0001.png From david.ponzone at gmail.com Mon Jun 7 10:34:12 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 7 Jun 2010 19:34:12 +0200 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: References: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> Message-ID: <70A84E11-A594-4534-8B98-5099A4F85EC2@gmail.com> Phillip, please do :) Well, I could be wrong, but this setup should not require any re-invite. I never really used bypass-media on FS, but from what I understood, it will jut advertise the customer IP to FS-RTP-3 and FS-RTP-3's IP to the customer. Anyway, the idea of the design was an attempt to answer to Code's question at the beginning of the thread. who wanted to build a such architecture with FS only. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 07/06/2010 ? 19:02, Phillip Jones a ?crit : > David, > > I hope you don't mind me interjecting here. But what is the > advantage of your setup over the traditional SIP Proxy - FS - SIP > Proxy setup? Isn't introducing a re-invite here, to shift the media > from FS-SIP-Internal-1 to FS-RTP-3 introducing a further > complication and potential point of failure? > > Pj > > On Mon, Jun 7, 2010 at 7:29 AM, David Ponzone > wrote: > Mike, > > You're right, it can be achieved with SIP now that I think a bit > more about it. > The idea was to allow adding multiple media gateways when required, > so the media gateways should not be facing the carriers as some of > them do SIP-filtering, but should only be advertised in the SDP. > > So SIP-only boxes (doing bypass-media) should face the carriers to > handle the trunking. > In the middle, we would then have the media gateways, doing SIP and > mostly RTP. > But I guess we dont want customers to register and to send calls to > a media gateway, so we need another set of SIP boxes on the other > side, doing bypass-media also. > > So it would like this: > > ------sip-----FS-RTP-1-----sip------ > FS-SIP-Internal-1 ------sip-----FS-RTP-2-----sip------FS-SIP- > External-1----sip-----Carriers > ------sip-----FS-RTP-3-----sip------ > FS-SIP-Internal-2 -------sip----FS-RTP-4-----sip------FS-SIP- > External-2-----sip----Carriers > -------sip----FS-RTP-5-----sip------ > > Thanks to bypass-media, the RTP streams would go from customer to FS- > RTP-x to Carriers, and reverse. > And I don't see any reason why the same set of FS-SIP boxes could > not be used for both internal and external borders. > > Is there something wrong in this ? > > Code, does it help ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 05/06/2010 ? 19:54, Michael Jerris a ?crit : > >> Why would it be an advantage to have your media proxies use another >> protocol? >> >> Mike >> >> On Jun 4, 2010, at 1:59 AM, David Ponzone wrote: >> >>> It doesn't solve the issue that all the media servers will do >>> signaling too, and will talk SIP with the carriers. >>> So the carriers will need to allow all the media servers . >>> >>> The only clean solution to avoid that, I think, is to have >>> signaling boxes allocating resources from media servers with >>> another protocol than SIP. >>> RTPproxy does that I think, but I am not sure how it works. >>> >>> David Ponzone >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/79aee0b9/attachment.html From sos at sokhapkin.dyndns.org Mon Jun 7 10:49:58 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 7 Jun 2010 13:49:58 -0400 Subject: [Freeswitch-users] RTP Proxy In-Reply-To: <1275931956.2807.26.camel@gustavo-laptop> References: <1275909314.2807.0.camel@gustavo-laptop> <1275931956.2807.26.camel@gustavo-laptop> Message-ID: <201006071349.59001.sos@sokhapkin.dyndns.org> As it was noted already, the best solution would be the combination of openser(load-balanser)->cluster_of_FS->openser(outgoing-proxy). Clients know only IP of load-balanser, termination gateways know only IP of outgoing-proxy. You can add/change nodes in FS cluster on the fly without any updates on clients or termination gateways end. Load balancer and outgoing proxy can be easily implemented in a single openser instance. On Monday 07 June 2010, Gustavo Espeche wrote: > basically i need split the sip signaling and sip media, one server with > FS for sip signaling and other one with proxy rtp. > i attached a png. > If some one work with FS in this schema, please let me know what > rtp-proxy do you use, this information can save to me lot of time > testing rtp-proxy. > > Regards. > > Gustavo Espeche > www.easyipcall.com > > On Mon, 2010-06-07 at 17:47 +0200, David Ponzone wrote: > > Perhaps can you explain exactly what you're trying to achieve. It > > would help the people here to help you :) > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > > ? l'intention exclusive de ses destinataires. Toute utilisation ou > > diffusion non autoris?e est interdite. Tout message ?lectronique est > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes > > pas destinataire de ce message, merci de le d?truire imm?diatement et > > d'avertir l'exp?diteur. > > > > Le 07/06/2010 ? 17:36, Gustavo Espeche a ?crit : > > > OK i know it, but i'm trying to configure an rtp-proxy in a > > > different > > > server that is running freeswitch. > > > > > > Gustavo Espeche > > > www.easyipcall.com > > > > > > On Mon, 2010-06-07 at 13:29 +0200, David Ponzone wrote: > > > > There is one included :) > > > > > > > > David Ponzone Direction Technique > > > > email: david.ponzone at ipeva.fr > > > > tel: 01 74 03 18 97 > > > > gsm: 06 66 98 76 34 > > > > > > > > > > > > Service Client IPeva > > > > tel: 0811 46 26 26 > > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > > ?tablis > > > > ? l'intention exclusive de ses destinataires. Toute utilisation ou > > > > diffusion non autoris?e est interdite. Tout message ?lectronique > > > > est > > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > > > titre > > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > > > n'?tes > > > > pas destinataire de ce message, merci de le d?truire imm?diatement > > > > et > > > > d'avertir l'exp?diteur. > > > > > > > > Le 07/06/2010 ? 13:15, Gustavo Espeche a ?crit : > > > > > Hi some one know an rtp proxy that work with freeswitch. > > > > > Thanks. > > > > > > > > > > Gustavo Espeche > > > > > www.easyipcall.com > > > > > > > > > > > > > > > _______________________________________________ > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > > > >users http://www.freeswitch.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-user > > >s http://www.freeswitch.org > From david.ponzone at gmail.com Mon Jun 7 10:51:52 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 7 Jun 2010 19:51:52 +0200 Subject: [Freeswitch-users] RTP Proxy In-Reply-To: <1275931956.2807.26.camel@gustavo-laptop> References: <1275909314.2807.0.camel@gustavo-laptop> <639452D0-B299-4214-8227-7CCBFAA8D49A@gmail.com> <1275924995.2807.9.camel@gustavo-laptop> <1275931956.2807.26.camel@gustavo-laptop> Message-ID: <229E0C9A-D87C-4ABA-B0D1-600588A4EF06@gmail.com> Gustavo, please read the recent thread called "FS as Media Gateway Only". It was about this exactly. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 07/06/2010 ? 19:32, Gustavo Espeche a ?crit : > > basically i need split the sip signaling and sip media, one server > with > FS for sip signaling and other one with proxy rtp. > i attached a png. > If some one work with FS in this schema, please let me know what > rtp-proxy do you use, this information can save to me lot of time > testing rtp-proxy. > > Regards. > > Gustavo Espeche > www.easyipcall.com > > On Mon, 2010-06-07 at 17:47 +0200, David Ponzone wrote: >> Perhaps can you explain exactly what you're trying to achieve. It >> would help the people here to help you :) >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis >> ? l'intention exclusive de ses destinataires. Toute utilisation ou >> diffusion non autoris?e est interdite. Tout message ?lectronique >> est >> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au >> titre >> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous >> n'?tes >> pas destinataire de ce message, merci de le d?truire imm?diatement >> et >> d'avertir l'exp?diteur. >> >> >> >> >> >> >> >> Le 07/06/2010 ? 17:36, Gustavo Espeche a ?crit : >> >>> OK i know it, but i'm trying to configure an rtp-proxy in a >>> different >>> server that is running freeswitch. >>> >>> Gustavo Espeche >>> www.easyipcall.com >>> >>> On Mon, 2010-06-07 at 13:29 +0200, David Ponzone wrote: >>>> There is one included :) >>>> >>>> David Ponzone Direction Technique >>>> email: david.ponzone at ipeva.fr >>>> tel: 01 74 03 18 97 >>>> gsm: 06 66 98 76 34 >>>> >>>> >>>> Service Client IPeva >>>> tel: 0811 46 26 26 >>>> www.ipeva.fr - www.ipeva-studio.com >>>> >>>> >>>> Ce message et toutes les pi?ces jointes sont confidentiels et >>>> ?tablis >>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou >>>> diffusion non autoris?e est interdite. Tout message ?lectronique >>>> est >>>> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au >>>> titre >>>> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si >>>> vous >>>> n'?tes >>>> pas destinataire de ce message, merci de le d?truire >>>> imm?diatement >>>> et >>>> d'avertir l'exp?diteur. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Le 07/06/2010 ? 13:15, Gustavo Espeche a ?crit : >>>> >>>>> Hi some one know an rtp proxy that work with freeswitch. >>>>> Thanks. >>>>> >>>>> Gustavo Espeche >>>>> www.easyipcall.com >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.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 >>> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/0fc77825/attachment-0001.html From msc at freeswitch.org Mon Jun 7 11:15:01 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 11:15:01 -0700 Subject: [Freeswitch-users] speech recognition? In-Reply-To: References: <7588B1B7A1B741A4A2A72C38249FC92D@dell9400> Message-ID: On Mon, Jun 7, 2010 at 5:34 AM, Myron Curtis wrote: > I will look more closely at the wiki to see if pcketsphinx + bridging > will do what I need. > > Thanks, > > Myron > Myron, Also check out Vestec. You can get an SDK for $25 and they have a FreeSWITCH connector. They also have really good recognition rates. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/fb5a2e60/attachment.html From msc at freeswitch.org Mon Jun 7 11:19:06 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 11:19:06 -0700 Subject: [Freeswitch-users] mod_conference Wiki - lists In-Reply-To: <1548359777.152.1275931350885.JavaMail.root@srvr12.remotelinkml.com> References: <1589313449.150.1275931207129.JavaMail.root@srvr12.remotelinkml.com> <1548359777.152.1275931350885.JavaMail.root@srvr12.remotelinkml.com> Message-ID: On Mon, Jun 7, 2010 at 10:22 AM, David Swardstrom < dswardstrom at remotelink.com> wrote: > I have been looking at the mod_conference Wiki page and have checked some > of it's information against the current code in mod_conference.c. In > particular, I have been interested in the ?list? capability which is > supported by conference_list(). The following comments will be used to > change the ?Mod Conference? Wiki page: > This applies to the API reference, list entry: > > > Output: First line > Conference ( member ) > Following lines are csv separated list for each conference leg with the > following items: > id of participiant (starts at 1 after FS restart) > Register string of participiant > Uuid of participiants call leg > Cid number > Cid name > Status (hear|speak|floor) > Volume In > Volume Out > Energy Level > > > > Output: First line > Conference ( member[s][ locked]) > Following lines are csv separated list for each conference leg with the > following items: > id of participant (starts at 1 after FS restart) > Register string of participant > Uuid of participants call leg > Caller id number > Caller id name > Status (hear|speak|talking|video|floor) > Volume In > Volume Out > Energy Level > Notes: > First line: ?locked? reflects the lock/unlock status of the conference. > Status: ?hear? reflects the mute/unmute status of the member. > Status: ?speak? reflects the ?deaf /undeaf? status of the member. > Status: ?talking? indicates that the input channel is providing some amount > of sound energy. > Status: ?video? ??? > Status: ?floor? indicates that this member currently owns the floor > > > Does anyone have any comments I should add for floor? Is there anything > significant to owning the floor? > How about video? It seems that video and video conferencing is a capability > that is desired and some support is provided. But is it really working? Does > anyone use it? Is there an assumption in conferencing that video is > currently sourced from only one place? > IIRC only one person can have the "floor" and when you're in a video conf the person who has the floor is displayed on all the clients. Someone please correct me if that is not the case. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/814e75f6/attachment.html From infos at madovsky.org Mon Jun 7 11:21:12 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Jun 2010 14:21:12 -0400 Subject: [Freeswitch-users] use variables set in sp_profiles/external/gateways Message-ID: <75EF468F7C33478D86E3BE7C3B9E98DC@MOBILEE1705> Hi, I'm trying to use the varaibles set in external gateway sip profiles in a dialplan with no success. I tried $${vars} and ${var} but the value is empty. Should I use them differently ? Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/1bbf60db/attachment.html From msc at freeswitch.org Mon Jun 7 11:21:37 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 11:21:37 -0700 Subject: [Freeswitch-users] Direct inward dialling In-Reply-To: References: <491B305B.4060307@gmx.net> <072A175C-9004-4C14-90EC-9A93A8453787@gmail.com> <35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org> Message-ID: On Mon, Jun 7, 2010 at 10:17 AM, stephen at stephenjc wrote: > i use this tool > > http://www.gskinner.com/RegExr/desktop/ > > I use these tools: fs_cli + regex command. ;) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/8e36e128/attachment.html From darkeagle6 at hotmail.com Mon Jun 7 11:22:35 2010 From: darkeagle6 at hotmail.com (Myron Curtis) Date: Mon, 7 Jun 2010 11:22:35 -0700 Subject: [Freeswitch-users] speech recognition? In-Reply-To: References: <36AC79320BB64F82B4E2A91ACA96B054@dell9400> Message-ID: Thanks! I think I can make this work, but it is going to be a challenge. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger Sent: Monday, June 07, 2010 6:55 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] speech recognition? I don't have any preferences. Anything that works goes - but I like the idea of a standard plug-in interface on this area. Most countries have speech detection, but you will not find a single vendor that supports all languages. In your case I would look for a natural speech engine - something I am not to familiar with myself. My last attempt on doing that gave some very interesting results J Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Myron Curtis Sent: 7. juni 2010 14:32 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] speech recognition? Thank you Jan, I t sounds like you might like a different speech recognition system for FreeSwitch better than Simon. Do you have suggestions for some that might be more compatible? I chose Simon only because it was recommended to me by a friend who has worked with it and he was willing to help set it up. Thanks for your quick reply. Myron From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger Sent: Monday, June 07, 2010 5:01 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] speech recognition? Myron, If you insist on using Simon I will suggest that you get some help from the Simon community to connect mrcp using unimrcp or similar onhttp://wiki.freeswitch.org/wiki/Mod_unimrcp. Once this is available it should just work. Simon is GPL - meaning the license is not compatible with FS - but, that does not prevent you as an end-user of using it. Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Myron Curtis Sent: 5. juni 2010 22:20 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] speech recognition? Hi, I am developing a virtual world called Virtual Worlds Grid, and I am using FreeSwitch to provide voice. I would like to integrate it with a speech recognition engine like Simon, in order to have everything said in voice to be typed into text. This is one way to make the grid accessible for people who cannot use voice, and it also allows language translation software like Open Bablefish, to translate voice conversations from the resulting text files, which would facilitate multilingual conversations. I am new to the FreeSwitch system, and need some guidance on how to set this up. I would appreciate any tips, cautions, or links to practical help I can get. If this will work in a virtual world, there is no reason it cannot be made to work on cell phones. Thanks, Myron Curtis No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.829 / Virus Database: 271.1.1/2923 - Release Date: 06/06/10 23:35:00 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.829 / Virus Database: 271.1.1/2923 - Release Date: 06/06/10 23:35:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/ffc1f091/attachment-0001.html From dswardstrom at remotelink.com Mon Jun 7 12:19:33 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Mon, 7 Jun 2010 14:19:33 -0500 (CDT) Subject: [Freeswitch-users] =?utf-8?q?mod=5Fconference_=E2=80=93_moderator?= =?utf-8?q?_flag_=26_xml=5Flist?= Message-ID: <1729271982.164.1275938373761.JavaMail.root@srvr12.remotelinkml.com> moderator flag Question: Is there any reason why mod_conference should be modified to provide the ?moderator? flag as part of the status on a list? I notice that it is provided by xml_list. Talking about xml_list. It does not seem to be documented and no caller within the Freeswitch C code. Is there any user? Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom From freeswitch at peely.com Mon Jun 7 12:22:23 2010 From: freeswitch at peely.com (peely) Date: Mon, 7 Jun 2010 12:22:23 -0700 (PDT) Subject: [Freeswitch-users] Change in behaviour for ESL events? In-Reply-To: <1275896373992-5147891.post@n2.nabble.com> References: <1275896373992-5147891.post@n2.nabble.com> Message-ID: <1275938543180-5150669.post@n2.nabble.com> Brilliant, working perfectly again, thanks! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Change-in-behaviour-for-ESL-events-tp5147891p5150669.html Sent from the freeswitch-users mailing list archive at Nabble.com. From infos at madovsky.org Mon Jun 7 12:24:19 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Jun 2010 15:24:19 -0400 Subject: [Freeswitch-users] vars scope in dialplan Message-ID: example : but ${prfx} is empty. I'm almost sure I didn't understand FS vars scope yet.... Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/e3abdc56/attachment.html From macedoslm at gmail.com Mon Jun 7 12:26:05 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Mon, 7 Jun 2010 16:26:05 -0300 Subject: [Freeswitch-users] Gateway Registration Message-ID: Hi, I want to use the same Gateway but with different users and passwords to make and receive calls. This is the scenario: I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch UserAgents (Sofia External) to this gateway so I can receive inbound calls. And to make a Outbound call I want to decide witch UserAgent I will use. Is there anyway to do this? Regards, -- Samuel Macedo Belo Horizonte - Brazil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/4ba035b4/attachment.html From macedoslm at gmail.com Mon Jun 7 12:31:25 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Mon, 7 Jun 2010 16:31:25 -0300 Subject: [Freeswitch-users] Gateway Registrations Message-ID: Hi, I want to use the same Gateway but with different users and passwords to make and receive calls. This is the scenario: I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch UserAgents (Sofia External) to this gateway so I can receive inbound calls. And to make a Outbound call I want to decide witch UserAgent I will use. Is there anyway to do this? Regards, -- Samuel Macedo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/aa9e4212/attachment.html From brian at freeswitch.org Mon Jun 7 12:32:41 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Jun 2010 14:32:41 -0500 Subject: [Freeswitch-users] vars scope in dialplan In-Reply-To: References: Message-ID: <506489B8-95F6-45C2-9785-1FDF2389D429@freeswitch.org> This is because variables are not SET like you think. See the dialplan is compiled before its executed... but you do have an option: The inline option will set it right then on the spot. /b On Jun 7, 2010, at 2:24 PM, Madovsky wrote: > example : > > > > > > > > > > > > > > but ${prfx} is empty. > I'm almost sure I didn't understand FS vars scope yet.... > > Thanks > > Franck From azatek0 at gmail.com Mon Jun 7 12:34:57 2010 From: azatek0 at gmail.com (Aza Tek) Date: Mon, 7 Jun 2010 21:34:57 +0200 Subject: [Freeswitch-users] simple fees bill In-Reply-To: References: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> <1915997343-1275875496-cardhu_decombobulator_blackberry.rim.net-298651853-@bda057.bisx.prodap.on.blackberry> Message-ID: Hi All I've looked everywhere for the Weekly Conference Recordings, where can I find them? Thanks On Mon, Jun 7, 2010 at 7:10 PM, Michael Collins wrote: > > FYI, > > Darren Schreiber is scheduled to talk about mod_nibblebill this coming > Wednesday on the FS community conf call: > http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_09 > > If you can't attend the call then look for the audio recording on the wiki > on Thursday. > -MC > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/399e8e57/attachment.html From anthony.minessale at gmail.com Mon Jun 7 12:38:01 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Jun 2010 14:38:01 -0500 Subject: [Freeswitch-users] Sending NOTIFY message to a phone via XML-RPC In-Reply-To: <4C0D24B0.3010708@gmx.net> References: <4C08C995.8060500@gmx.net> <4C0D24B0.3010708@gmx.net> Message-ID: you can't do sendevent over XML its a command for event_socket it would probably be possible to make such a thing but it does not currently exist. On Mon, Jun 7, 2010 at 11:56 AM, Peter P GMX wrote: > Nobody has a working XML sample for sending NOTIFY events? > The wiki is quite lean on this. I will update the wiki with a working > sample as soon as I have one. > > So any suggestions? > > Best regards > Peter > > Peter P GMX schrieb: > > Hello, > > > > I try to send a NOTIFY message from my application to a registered phone > > via XML-RPC. > > This is the XML which is sent: > > > > > > > > freeswitch.api > > > > > > > > sendevent > > > > > > > > > > > NOTIFY,profile=internal,event-string=check-sync;reboot=false,user=200,host=192.168.178.220,content-type=application/simple-message-summary > > > > > > > > > > > > However I receive an error message: > > . > > . > > . > > ERROR!. > > . > > . > > > > I also tried to send the data as a struct but FS complains about needing > > a string as parameters. > > > > What am I doing wrong? Anybody has a valid sample XML for a NOTIFY > message? > > > > Best regards > > Peter > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > -- 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/20100607/b7eac773/attachment.html From anthony.minessale at gmail.com Mon Jun 7 12:38:49 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Jun 2010 14:38:49 -0500 Subject: [Freeswitch-users] Gateway Registration In-Reply-To: References: Message-ID: Hi, I would like to receive only one copy of any given email. On Mon, Jun 7, 2010 at 2:26 PM, Samuel Macedo wrote: > Hi, > > I want to use the same Gateway but with different users and passwords to > make and receive calls. > This is the scenario: > I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch UserAgents > (Sofia External) to this gateway so I can receive inbound calls. And to make > a Outbound call I want to decide witch UserAgent I will use. > > Is there anyway to do this? > > Regards, > -- > Samuel Macedo > Belo Horizonte - Brazil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100607/fbeb26be/attachment-0001.html From anthony.minessale at gmail.com Mon Jun 7 12:39:01 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Jun 2010 14:39:01 -0500 Subject: [Freeswitch-users] Gateway Registrations In-Reply-To: References: Message-ID: Hi, I would like to receive only one copy of any given email. On Mon, Jun 7, 2010 at 2:31 PM, Samuel Macedo wrote: > Hi, > > I want to use the same Gateway but with different users and passwords to > make and receive calls. > This is the scenario: > I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch UserAgents > (Sofia External) to this gateway so I can receive inbound calls. And to make > a Outbound call I want to decide witch UserAgent I will use. > > Is there anyway to do this? > > Regards, > -- > Samuel Macedo > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100607/52ed3f35/attachment.html From infos at madovsky.org Mon Jun 7 12:44:54 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Jun 2010 15:44:54 -0400 Subject: [Freeswitch-users] vars scope in dialplan Message-ID: <60537C59088D495EAA0FDEF3A77B29FF@MOBILEE1705> Ok I found the answer... http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12449.html F ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 07, 2010 3:24 PM Subject: vars scope in dialplan example : but ${prfx} is empty. I'm almost sure I didn't understand FS vars scope yet.... Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/fa55b3e6/attachment.html From darkeagle6 at hotmail.com Mon Jun 7 12:47:26 2010 From: darkeagle6 at hotmail.com (Myron Curtis) Date: Mon, 7 Jun 2010 12:47:26 -0700 Subject: [Freeswitch-users] speech recognition? In-Reply-To: References: <7588B1B7A1B741A4A2A72C38249FC92D@dell9400> Message-ID: I had not heard of them until now, thanks. This quest is going to open a whole new universe for me to conquer. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, June 07, 2010 11:15 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] speech recognition? On Mon, Jun 7, 2010 at 5:34 AM, Myron Curtis wrote: I will look more closely at the wiki to see if pcketsphinx + bridging will do what I need. Thanks, Myron Myron, Also check out Vestec. You can get an SDK for $25 and they have a FreeSWITCH connector. They also have really good recognition rates. -MC No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.829 / Virus Database: 271.1.1/2923 - Release Date: 06/06/10 23:35:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/9f13a3d2/attachment.html From infos at madovsky.org Mon Jun 7 12:53:57 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Jun 2010 15:53:57 -0400 Subject: [Freeswitch-users] vars scope in dialplan Message-ID: <62D8A8CE8DA846219FDF13CB83933370@MOBILEE1705> wrong info... it doesn't work also... anyone ? Thanks F ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 07, 2010 3:44 PM Subject: Re: vars scope in dialplan Ok I found the answer... http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12449.html F ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 07, 2010 3:24 PM Subject: vars scope in dialplan example : but ${prfx} is empty. I'm almost sure I didn't understand FS vars scope yet.... Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/25634012/attachment.html From infos at madovsky.org Mon Jun 7 12:55:46 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Jun 2010 15:55:46 -0400 Subject: [Freeswitch-users] vars scope in dialplan References: <506489B8-95F6-45C2-9785-1FDF2389D429@freeswitch.org> Message-ID: Thanks Brian, I forgot this option... F ----- Original Message ----- From: "Brian West" To: Sent: Monday, June 07, 2010 3:32 PM Subject: Re: [Freeswitch-users] vars scope in dialplan > This is because variables are not SET like you think. See the dialplan is > compiled before its executed... but you do have an option: > > > > > The inline option will set it right then on the spot. > > /b > > > On Jun 7, 2010, at 2:24 PM, Madovsky wrote: > >> example : >> >> >> >> >> >> >> >> >> >> >> >> >> >> but ${prfx} is empty. >> I'm almost sure I didn't understand FS vars scope yet.... >> >> Thanks >> >> Franck > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Mon Jun 7 14:05:26 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 14:05:26 -0700 Subject: [Freeswitch-users] simple fees bill In-Reply-To: References: <14500BC566B24B80A9CD67FC8F8A4D3E@MOBILEE1705> <1915997343-1275875496-cardhu_decombobulator_blackberry.rim.net-298651853-@bda057.bisx.prodap.on.blackberry> Message-ID: On Mon, Jun 7, 2010 at 12:34 PM, Aza Tek wrote: > Hi All > > I've looked everywhere for the Weekly Conference Recordings, where can I > find them? > > They will be linked from the weekly conference agenda page as well as on the wiki page of the subject matter being discussed. The mod_nibblebill presentation has not happened yet so there is no recording. You can see links to past recordings by visiting the specific agenda pages which are indexed here: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/1d079108/attachment-0001.html From infos at madovsky.org Mon Jun 7 14:06:16 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Jun 2010 17:06:16 -0400 Subject: [Freeswitch-users] vars scope in dialplan References: <506489B8-95F6-45C2-9785-1FDF2389D429@freeswitch.org> Message-ID: Brien, if I have to write several conditions for the same vars, like ${prfx} is empty. is not allowed to do that ? Thanks Franck ----- Original Message ----- From: "Madovsky" To: Sent: Monday, June 07, 2010 3:55 PM Subject: Re: [Freeswitch-users] vars scope in dialplan > Thanks Brian, I forgot this option... > > F > > ----- Original Message ----- > From: "Brian West" > To: > Sent: Monday, June 07, 2010 3:32 PM > Subject: Re: [Freeswitch-users] vars scope in dialplan > > >> This is because variables are not SET like you think. See the dialplan >> is >> compiled before its executed... but you do have an option: >> >> >> >> >> The inline option will set it right then on the spot. >> >> /b >> >> >> On Jun 7, 2010, at 2:24 PM, Madovsky wrote: >> >>> example : >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> but ${prfx} is empty. >>> I'm almost sure I didn't understand FS vars scope yet.... >>> >>> Thanks >>> >>> Franck >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 From codeghar at gmail.com Mon Jun 7 14:31:34 2010 From: codeghar at gmail.com (Code Ghar) Date: Mon, 7 Jun 2010 16:31:34 -0500 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: <70A84E11-A594-4534-8B98-5099A4F85EC2@gmail.com> References: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> <70A84E11-A594-4534-8B98-5099A4F85EC2@gmail.com> Message-ID: I haven't used bypass-media extensively, either. So I am also interested in knowing if it would introduce re-invite. I know several carriers, using some older SIP implementations or for other reasons, do not support re-invite. In a perfect world we would use bypass-media on the ingress and egress FS-SIP servers, without introducing re-invite, and handle media using intermediate FS-RTP servers. Aside from the media, if we use, say two FS-SIP servers, then all FS-RTP servers can do load balance when doing egress. In this way we only need two signaling IPs for all kinds of customers and carriers and don't have to differentiate between ingress-only and egress-only FS-SIP servers. Each FS-SIP is ingress and egress while all FS-RTP look only like media IPs to all customers and carriers. Of course, I would like to get consensus confirmation from experienced users that bypass-media does not introduce re-invite. We could use a SIP Proxy, as advised by PJ, but if a bunch of FS servers could do that same job that would be awesome, too. On Mon, Jun 7, 2010 at 12:34 PM, David Ponzone wrote: > Phillip, > > please do :) > > Well, I could be wrong, but this setup should not require any re-invite. > I never really used bypass-media on FS, but from what I understood, it will > jut advertise the customer IP to FS-RTP-3 and FS-RTP-3's IP to the customer. > > Anyway, the idea of the design was an attempt to answer to Code's question > at the beginning of the thread. who wanted to build a such architecture with > FS only. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 07/06/2010 ? 19:02, Phillip Jones a ?crit : > > David, > > I hope you don't mind me interjecting here. But what is the advantage of > your setup over the traditional SIP Proxy - FS - SIP Proxy setup? Isn't > introducing a re-invite here, to shift the media from FS-SIP-Internal-1 to > FS-RTP-3 introducing a further complication and potential point of failure? > > Pj > > On Mon, Jun 7, 2010 at 7:29 AM, David Ponzone wrote: > >> Mike, >> >> You're right, it can be achieved with SIP now that I think a bit more >> about it. >> The idea was to allow adding multiple media gateways when required, so the >> media gateways should not be facing the carriers as some of them do >> SIP-filtering, but should only be advertised in the SDP. >> >> So SIP-only boxes (doing bypass-media) should face the carriers to handle >> the trunking. >> In the middle, we would then have the media gateways, doing SIP and mostly >> RTP. >> But I guess we dont want customers to register and to send calls to a >> media gateway, so we need another set of SIP boxes on the other side, doing >> bypass-media also. >> >> So it would like this: >> >> ------sip-----FS-RTP-1-----sip------ >> FS-SIP-Internal-1 >> ------sip-----FS-RTP-2-----sip------FS-SIP-External-1----sip-----Carriers >> ------sip-----FS-RTP-3-----sip------ >> FS-SIP-Internal-2 >> -------sip----FS-RTP-4-----sip------FS-SIP-External-2-----sip----Carriers >> -------sip----FS-RTP-5-----sip------ >> >> Thanks to bypass-media, the RTP streams would go from customer to FS-RTP-x >> to Carriers, and reverse. >> And I don't see any reason why the same set of FS-SIP boxes could not be >> used for both internal and external borders. >> >> Is there something wrong in this ? >> >> Code, does it help ? >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur.* >> * >> * >> >> >> >> Le 05/06/2010 ? 19:54, Michael Jerris a ?crit : >> >> Why would it be an advantage to have your media proxies use another >> protocol? >> >> Mike >> >> On Jun 4, 2010, at 1:59 AM, David Ponzone wrote: >> >> It doesn't solve the issue that all the media servers will do signaling >> too, and will talk SIP with the carriers. >> So the carriers will need to allow all the media servers . >> >> The only clean solution to avoid that, I think, is to have signaling boxes >> allocating resources from media servers with another protocol than SIP. >> RTPproxy does that I think, but I am not sure how it works. >> >> David Ponzone >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100607/de407b42/attachment.html From codeghar at gmail.com Mon Jun 7 14:37:39 2010 From: codeghar at gmail.com (Code Ghar) Date: Mon, 7 Jun 2010 16:37:39 -0500 Subject: [Freeswitch-users] dynamic dial plan with lua In-Reply-To: References: Message-ID: I am also interested in an answer to this question. But since I am more proficient in Python, I would ask the same using Python instead of Lua. But Tony I think if we can get your question answered we can probably assume it would work with Python, Perl, etc. If you are interested, I have had success with using mod_xml_curl and a prototype application written in Django. It basically reads the POST data and then returns an XML response which is the dial plan. But if the same could be done using simple Lua or Python or whatever, it would be great. On Mon, Jun 7, 2010 at 12:41 AM, Tony Tin wrote: > Hi, > > I'm trying to implement dynamic dial plan, my approach is calling a lua > scrip in "default.xml" as below, so I can change the dial plan by changing > the lua script at any time. > > > > Could any on please give me some advice about my approach and whether there > is any limitation of LUA regarding dial plan implemenation. > > Thanks. > > Tony > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/b0dd97e1/attachment-0001.html From brian at freeswitch.org Mon Jun 7 14:42:24 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Jun 2010 16:42:24 -0500 Subject: [Freeswitch-users] dynamic dial plan with lua In-Reply-To: References: Message-ID: <78D036C2-F564-4F32-A34C-357B2B84C7F9@freeswitch.org> http://wiki.freeswitch.org/wiki/Serve_configs_like_xml_curl I take it you didn't read the mod_lua wiki page? /b On Jun 7, 2010, at 4:37 PM, Code Ghar wrote: > I am also interested in an answer to this question. But since I am more proficient in Python, I would ask the same using Python instead of Lua. But Tony I think if we can get your question answered we can probably assume it would work with Python, Perl, etc. > > If you are interested, I have had success with using mod_xml_curl and a prototype application written in Django. It basically reads the POST data and then returns an XML response which is the dial plan. But if the same could be done using simple Lua or Python or whatever, it would be great. From msc at freeswitch.org Mon Jun 7 14:51:08 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 14:51:08 -0700 Subject: [Freeswitch-users] Freeswitch download and configuration Setup webpage In-Reply-To: References: <479303.50690.qm@web43403.mail.sp1.yahoo.com> Message-ID: Let's not forget the noob article: http://bit.ly/EpVrv -MC On Fri, Jun 4, 2010 at 10:43 PM, Wasim Baig wrote: > http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ > > http://wiki.freeswitch.org/wiki/Download_FreeSWITCH > > http://wiki.freeswitch.org/wiki/Installation_Guide > > -waism > > > On Sat, Jun 5, 2010 at 10:32, Shaik basha wrote: > >> >> Dear all, >> >> Good morning every one. I a m a new bie in freeswitch, though I tried to >> search the download page and configuration set up. but, I don't see any >> where. Can any one help me in this regard. I have spent several hours, >> though I was not succeeded. >> >> Hence I kindly request to let me know from where I can download and how to >> do configuration setup. Thanking in advance. earliest response in this >> regard would be very much appreciated. I would be very thankful and grateful >> for your kind information. Regards, >> >> shaikbashaatc >> +919246769086 >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 | > peace be upon you ... > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/55356cc7/attachment.html From codeghar at gmail.com Mon Jun 7 14:52:28 2010 From: codeghar at gmail.com (Code Ghar) Date: Mon, 7 Jun 2010 16:52:28 -0500 Subject: [Freeswitch-users] Convert DTMF Between Inband and Out-of-band Message-ID: I have come across a scenario: server (proprietary software, etc. -- let's call it S1) receives a new call from ingress carrier (let's call it IN1), does its thing, and connects to egress carrier (let's call it OUT1). Once call is answered by egress end, it does a re-invite between ingress and egress carriers. This re-invite keeps the server (S1) in signaling path but makes the ingress (IN1) and egress (OUT1) carriers send RTP to each other directly. Initial negotiation does DTMF using RFC2833 (out-of-band) but still in the media stream. The result is that after re-invite if IN1 sends a DTMF event meant for S1, it actually goes to OUT1 instead. One solution is to use SIP INFO for DTMF. Since S1 is in signaling path it will receive all SIP INFO packets and do what with them what needs to be done. And this is where my question arises. Let's say our FS server is IN1. It also has an ingress and egress carrier. In this case ingress could be anyone (let's call it INORIG) but egress is S1. If IN1 receives DTMF event using RFC2833 but needs to send the event to S1 using SIP INFO, can it be done in FreeSWITCH? In effect, can FS do conversion between RFC2833 and SIP INFO? And if it can, can this conversion be used for only some gateways and not all? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/83672ecd/attachment.html From lon at kickasspixels.com Mon Jun 7 14:52:29 2010 From: lon at kickasspixels.com (Kickass Pixels) Date: Mon, 7 Jun 2010 14:52:29 -0700 Subject: [Freeswitch-users] =?windows-1252?q?mod=5Fconference_=96_moderato?= =?windows-1252?q?r_flag_=26_xml=5Flist?= In-Reply-To: <1729271982.164.1275938373761.JavaMail.root@srvr12.remotelinkml.com> References: <1729271982.164.1275938373761.JavaMail.root@srvr12.remotelinkml.com> Message-ID: David, I submitted a patch months ago to provide more of the existing flags in both the xml_list and through the event socket. This information makes it trivial to write monitoring and event socket applications to manage conferences. The moderator flag is a crucial flag for differentiating rights. -- Lon On Jun 7, 2010, at 12:19 PM, David Swardstrom wrote: > moderator flag > Question: Is there any reason why mod_conference should be modified to provide the ?moderator? flag as part of the status on a list? I notice that it is provided by xml_list. > > Talking about xml_list. It does not seem to be documented and no caller within the Freeswitch C code. Is there any user? > > Regards, > David Swardstrom > (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Mon Jun 7 14:58:36 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 14:58:36 -0700 Subject: [Freeswitch-users] javascript bridge In-Reply-To: References: Message-ID: I take it that this js is being called from the dialplan? If so then do what you need to do with outsession before you bridge it. Or consider using ESL and making a socket-based control program where you have much more flexibility... -MC On Sun, Jun 6, 2010 at 11:55 AM, stephen at stephenjc wrote: > I have the following code, and after the bridge the javascript seems to > stop. > > outsession = new > Session("{ringback=\'%(2000,4000,440.0,480.0)\',instant_ringback=true,ignore_early_media=false}sofia/gateway/" > + providerhash["providername0"] + "/XXXXXXXXXXXX"); > bridge(session,outsession); > while(outsession.ready()) > { > console_log("notice","ping"); > } > > I am looking to manage the b leg, should i use execute_on_answer instead? > or is there a way to make the code continue after the bridge. > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/e61ec8f8/attachment.html From msc at freeswitch.org Mon Jun 7 15:17:59 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jun 2010 15:17:59 -0700 Subject: [Freeswitch-users] vars scope in dialplan In-Reply-To: References: <506489B8-95F6-45C2-9785-1FDF2389D429@freeswitch.org> Message-ID: On Mon, Jun 7, 2010 at 2:06 PM, Madovsky wrote: > Brien, > > if I have to write several conditions for the same vars, like > > > > > > > > > > > > ${prfx} is empty. is not allowed to do that ? > > At what point is ${prfx} empty? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/3e8990e5/attachment.html From infos at madovsky.org Mon Jun 7 18:25:45 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Jun 2010 21:25:45 -0400 Subject: [Freeswitch-users] vars scope in dialplan References: <506489B8-95F6-45C2-9785-1FDF2389D429@freeswitch.org> Message-ID: <6CE8FD65D419416BAA678A6F0E8BCE6B@MOBILEE1705> sorry if i put another condition after this block like at the last condition block i${prfx} is empty Thanks Franck ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 07, 2010 6:17 PM Subject: Re: [Freeswitch-users] vars scope in dialplan On Mon, Jun 7, 2010 at 2:06 PM, Madovsky wrote: Brien, if I have to write several conditions for the same vars, like ${prfx} is empty. is not allowed to do that ? At what point is ${prfx} empty? -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/2e76ebd1/attachment-0001.html From brian at freeswitch.org Mon Jun 7 18:31:21 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Jun 2010 20:31:21 -0500 Subject: [Freeswitch-users] vars scope in dialplan In-Reply-To: <6CE8FD65D419416BAA678A6F0E8BCE6B@MOBILEE1705> References: <506489B8-95F6-45C2-9785-1FDF2389D429@freeswitch.org> <6CE8FD65D419416BAA678A6F0E8BCE6B@MOBILEE1705> Message-ID: sounds like ${cake} didn't exist. Did you eat it already? /b On Jun 7, 2010, at 8:25 PM, Madovsky wrote: > sorry > if i put another condition after this block like > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/8f4ecc71/attachment.html From infos at madovsky.org Mon Jun 7 18:42:59 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Jun 2010 21:42:59 -0400 Subject: [Freeswitch-users] vars scope in dialplan References: <506489B8-95F6-45C2-9785-1FDF2389D429@freeswitch.org><6CE8FD65D419416BAA678A6F0E8BCE6B@MOBILEE1705> Message-ID: No, maybe dialplan ;) ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 07, 2010 9:31 PM Subject: Re: [Freeswitch-users] vars scope in dialplan sounds like ${cake} didn't exist. Did you eat it already? /b On Jun 7, 2010, at 8:25 PM, Madovsky wrote: sorry if i put another condition after this block like ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/39e704d7/attachment.html From infos at madovsky.org Mon Jun 7 19:04:25 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 7 Jun 2010 22:04:25 -0400 Subject: [Freeswitch-users] vars scope in dialplan Message-ID: <3A4E4E9257154BC39E4BFE58BD165F62@MOBILEE1705> if I understand multiple conditions in an extension mean "if cond1 AND cond2 THEN condLAST" ? if yes, how to use condtions as IF cond1 OR cond2 the condLAST ? Thanks ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 07, 2010 9:31 PM Subject: Re: [Freeswitch-users] vars scope in dialplan sounds like ${cake} didn't exist. Did you eat it already? /b On Jun 7, 2010, at 8:25 PM, Madovsky wrote: sorry if i put another condition after this block like ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100607/a572a273/attachment.html From yehavi.bourvine at gmail.com Mon Jun 7 21:27:02 2010 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Jun 2010 07:27:02 +0300 Subject: [Freeswitch-users] Convert DTMF Between Inband and Out-of-band In-Reply-To: References: Message-ID: If you can use different profiles for ingress and egress then simply define the correct signalling on each profile. That's the way I do it here (Polycom phones which understand only RFC-2833 trying to talk with a Cisco gateay that has a bug with its 2833...). __Yehavi: 2010/6/8 Code Ghar > I have come across a scenario: server (proprietary software, etc. -- let's > call it S1) receives a new call from ingress carrier (let's call it IN1), > does its thing, and connects to egress carrier (let's call it OUT1). Once > call is answered by egress end, it does a re-invite between ingress and > egress carriers. This re-invite keeps the server (S1) in signaling path but > makes the ingress (IN1) and egress (OUT1) carriers send RTP to each other > directly. Initial negotiation does DTMF using RFC2833 (out-of-band) but > still in the media stream. The result is that after re-invite if IN1 sends a > DTMF event meant for S1, it actually goes to OUT1 instead. > > One solution is to use SIP INFO for DTMF. Since S1 is in signaling path it > will receive all SIP INFO packets and do what with them what needs to be > done. And this is where my question arises. Let's say our FS server is IN1. > It also has an ingress and egress carrier. In this case ingress could be > anyone (let's call it INORIG) but egress is S1. If IN1 receives DTMF event > using RFC2833 but needs to send the event to S1 using SIP INFO, can it be > done in FreeSWITCH? In effect, can FS do conversion between RFC2833 and SIP > INFO? And if it can, can this conversion be used for only some gateways and > not all? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/ff034551/attachment.html From thangappan143 at gmail.com Mon Jun 7 21:45:12 2010 From: thangappan143 at gmail.com (Thangappan.M) Date: Tue, 8 Jun 2010 10:15:12 +0530 Subject: [Freeswitch-users] Need to stop more than one voice file using break application In-Reply-To: References: Message-ID: In async mode set the event lock true. When the playback is going, just in the program waited for the playback has to be completed. While playback is going, passed the signals to this process. At that time I need to stop the current playback. In the signal handler using " break all" stop the playback but it was not stopping the playback. If I do the " break all" in the normal place( not the signal handler) it is being worked fine. On Mon, Jun 7, 2010 at 10:38 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I didn't understand you. > > > On Mon, Jun 7, 2010 at 6:48 AM, Thangappan.M wrote: > >> For normal scenario it is working fine. I have done the break application >> in the signal handler at that time it is not working. >> >> Any reason? >> >> >> >> >> On Fri, Jun 4, 2010 at 9:03 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> I added a patch [eba05c3 ] so it will work now >>> >>> >>> On Fri, Jun 4, 2010 at 4:01 AM, Thangappan.M wrote: >>> >>>> >>>> Dear all, >>>> >>>> I am in the process of developing IVR using FreeSWITCH. For that I >>>> am being used outbound ESL in async mode. >>>> >>>> In my design, usually for one menu it might be more than one voice >>>> files. So using playback_delimiter, play back all the voice file in single >>>> instance using playback application. >>>> >>>> I've tried to stop the play back using "break all" API. But it >>>> only break the only one voice file not the whole application(playback). >>>> >>>> Consider that I am playback four voice files. When the first voice >>>> file is getting playback, using " break all" t stop the playback . >>>> It only stopped the second voice and continued to playback the third voice >>>> file and followed by fourth one. >>>> >>>> My need is to break the playback application which may have any >>>> number of voice files in async mode. >>>> >>>> Thanks in advance. >>>> >>>> >>>> -- >>>> Regards, >>>> Thangappan.M >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Regards, >> Thangappan.M >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/8f366a77/attachment-0001.html From rogelio.perez at gmail.com Mon Jun 7 23:32:54 2010 From: rogelio.perez at gmail.com (Rogelio Perez) Date: Tue, 8 Jun 2010 03:32:54 -0300 Subject: [Freeswitch-users] Proxy Media for IVR bridged calls Message-ID: <0E902AFE-B255-4A9B-8DA2-27F3347390DC@gmail.com> Hi all, I have configured my internal profile to proxy all media, so I can use the g.729 codec for calls between extensions, even when they are behind NAT. Now I have a DID pointing to my FS instance, which triggers a standard IVR with a greeting message and a directory. Then someone calls the IVR and dial any extension, and when the call is answered it fails with error: error mod_g729.c:102 This codec is only usable in passthrough mode! I believe that the IVR bridge is not using proxy media, and that's why the call fails. My question is: how do I force proxy media for IVR bridged calls? Thanks! Rogelio From babak.freeswitch at gmail.com Tue Jun 8 00:08:06 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Tue, 8 Jun 2010 11:38:06 +0430 Subject: [Freeswitch-users] odbc on windows In-Reply-To: References: Message-ID: no one used ODBC on windows? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/55b3b905/attachment.html From peter.olsson at visionutveckling.se Tue Jun 8 00:09:21 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 8 Jun 2010 09:09:21 +0200 Subject: [Freeswitch-users] CHANNEL_PARK event not triggered anymore from event socket? Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C567CC7AECC@cooper> I'm not filing this as a bug right now, since I don't know if this was an intended change. I used to trigger on the CHANNEL_PARK event, but it doesn't seem that this event is fired anymore. Is this change intended? If not, I will submit a jira report with full logs showing this issue. /Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/e5b4292f/attachment.html From peter.olsson at visionutveckling.se Tue Jun 8 00:17:04 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 8 Jun 2010 09:17:04 +0200 Subject: [Freeswitch-users] odbc on windows In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C567CC7AEDD@cooper> I've only used it from javascript (spidermonkey), and there it works as expected. But I didn't try to use odbc for the core db (yet). /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r babak yakhchali Skickat: den 8 juni 2010 09:08 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] odbc on windows no one used ODBC on windows? !DSPAM:4c0dee0732931407819578! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/a6f1138f/attachment.html From david.ponzone at gmail.com Tue Jun 8 00:28:44 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Tue, 8 Jun 2010 09:28:44 +0200 Subject: [Freeswitch-users] Proxy Media for IVR bridged calls In-Reply-To: <0E902AFE-B255-4A9B-8DA2-27F3347390DC@gmail.com> References: <0E902AFE-B255-4A9B-8DA2-27F3347390DC@gmail.com> Message-ID: Rogelio, proxy-media is not required for G729 passthrough. By default, FS does relay media, and G729, among others, is available as a passthrough pseudo-codec. Proxy-media tells it to be a little bit more transparent, so you can use codecs that FS does not handle at all, or you can relay T38. In your case, I think your issue is a codec negotiation one. If a call from the outside hits your IVR, perhaps G711 is negotiated (that depends on the order of the codecs proposed by your carrier). And if you are using the default IVR sound files, I am pretty sure it was G729, because if not, you wouldn't be able to hear anything. Then, if you bridge to a G729 phone, you are asking FS to transcode between leg A, which was G711 and leg B which is G729. What you have to do is: -force leg A to be G729 -convert all your IVR sound files to G729, make sure FS loads mod_native_file and in your IVR config, reference all sound files without their extension David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 08/06/2010 ? 08:32, Rogelio Perez a ?crit : > Hi all, > > I have configured my internal profile to proxy all media, so I can > use the g.729 codec for calls between extensions, even when they are > behind NAT. > Now I have a DID pointing to my FS instance, which triggers a > standard IVR with a greeting message and a directory. > Then someone calls the IVR and dial any extension, and when the call > is answered it fails with error: error mod_g729.c:102 This codec is > only usable in passthrough mode! > I believe that the IVR bridge is not using proxy media, and that's > why the call fails. > > My question is: how do I force proxy media for IVR bridged calls? > > Thanks! > Rogelio > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/f20da17b/attachment-0001.html From vfclists at gmail.com Tue Jun 8 00:33:50 2010 From: vfclists at gmail.com (Frank Church) Date: Tue, 8 Jun 2010 08:33:50 +0100 Subject: [Freeswitch-users] odbc on windows In-Reply-To: References: Message-ID: You are not the only one with this question, I hope I will get the chance to make a few tests next week. On 8 June 2010 08:08, babak yakhchali wrote: > no one used ODBC on windows? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Frank Church ======================= http://devblog.brahmancreations.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/f846cec9/attachment.html From lists at infosecurity.ch Tue Jun 8 01:37:30 2010 From: lists at infosecurity.ch (Fabio Pietrosanti (naif)) Date: Tue, 08 Jun 2010 10:37:30 +0200 Subject: [Freeswitch-users] Full NAT bypass solution with STUN/ICE/FS but without TURN: is possible? Message-ID: <4C0E014A.1000507@infosecurity.ch> Hi all, i need to setup a server that handle VoIP connection in a way that always works in the most efficient way respect to latency and NAT traversal. FS provide very good NAT detection system but unfortunately when it fail, it just fail and does not have any kind of fallback mechanism. If FS do bypass-media and the client cannot communicate because the phone call is lost. If FS do proxy-media, the clients can always communicate (over UDP) but there is added latency. So we cannot rely only on FS NAT detection system and need also some client side system. Now, in theory the best approach would be not to touch FS but just use client-side system based on ICE/STUN with a TURN server-side component as a fallback. However there's no widely available TURN server, or at least TURN is not widely and well diffused just now. I am wondering whether it may be possible to make a setup where: - Client try to establish peer to peer connectivity with STUN/ICE (so FS is doing bypass-media not touching the SDP) - If they cannot work, will go in fallback to FS (but with proxy-media) instead than going via TURN So using the FS B2BUA proxy-media feature as alternative to TURN server, but letting clients to dynamically try to connect directly one each other by using the ICE methods that are dynamic. Does it seems something feasible? Fabio Pietrosanti From Prometheus001 at gmx.net Tue Jun 8 01:44:29 2010 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 08 Jun 2010 10:44:29 +0200 Subject: [Freeswitch-users] Messaging: Content-Type: application/octet-stream changed to text/html. Message-ID: <4C0E02ED.9000204@gmx.net> Hello, we have a SIP based Videoconferencing solution which we would like to integrate in Freeswitch. Registering the endpoints works, however all other operations which are done via messaging do not work as expected, as Freeswitch changes the Content-Type, see below. How can we tell Freeswitch not to change the Content-Type? Best regards Peter Original message from UA to Freeswitch with "Content-Type: application/octet-stream.": U 2010/06/01 18:21:57.056468 192.168.178.145:5530 -> 192.168.178.220:5060 MESSAGE sip:835351 at my.domain SIP/2.0. From: Hans ;tag=de58b276076cdf119f8bcf0221b28bff. To: sip:835351 at my.domain. Via: SIP/2.0/UDP 192.168.178.145:5530;branch=z9hG4bKae56b276076cdf119f8bcf0221b28bff;uas-addr=192.168.178.220. CSeq: 3 MESSAGE. Call-ID: 645ab276-076c-df11-824f-cf0221b28bff. Contact: "Hans". User-Agent: BRAVIS/0.0.0.27.4675 (Linux 2.6.32-22-generic; generic; Ubuntu 10.04 LTS; i686; de; 8). Content-Type: application/octet-stream. Content-Length: 231. . BRVSDRABAAEAAACeAAAAGfLjfQQAAAAvrRdulaIBAAAAMQEA/wAASwUAAABzcmZseP////+n pgi9D7MvrRduD7MBAAAANAEA/QAASwUAAABzcmZseP////+npgi9D7Onpgi9D7MBAAAAMwEA BVHDfgQAAABob3N0/////5WmD7d9pQQUp3cPswEAAAAyAQADUcN+BAAAAGhvc3QA////laYP t32lAgAAAA==. Modified message from Freeswitch to another UA with "Content-Type: text/html": U 2010/06/01 18:21:57.061965 192.168.178.220:5060 -> 192.168.178.122:5530 MESSAGE sip:835351 at 192.168.178.122:5530;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.178.220;rport;branch=z9hG4bKaHZUBrU7S2BXQ. Max-Forwards: 70. From: Hans ;tag=de58b276076cdf119f8bcf0221b28bff. To: "Peter ". Call-ID: a3bd6fee-e83c-122d-1ca4-080027e51f59. CSeq: 131586180 MESSAGE. Contact:. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Content-Type: text/html. Content-Length: 233. . BRVSDRABAAEAAACeAAAAGfLjfQQAAAAvrRdulaIBAAAAMQEA/wAASwUAAABzcmZseP////+n pgi9D7MvrRduD7MBAAAANAEA/QAASwUAAABzcmZseP////+npgi9D7Onpgi9D7MBAAAAMwEA BVHDfgQAAABob3N0/////5WmD7d9pQQUp3cPswEAAAAyAQADUcN+BAAAAGhvc3QA////laYP t32lAgAAAA==. From david.ponzone at gmail.com Tue Jun 8 01:53:16 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Tue, 8 Jun 2010 10:53:16 +0200 Subject: [Freeswitch-users] Full NAT bypass solution with STUN/ICE/FS but without TURN: is possible? In-Reply-To: <4C0E014A.1000507@infosecurity.ch> References: <4C0E014A.1000507@infosecurity.ch> Message-ID: <31EED764-8AAD-4CE8-98BD-E6CE44F80162@gmail.com> Fabio, this sounds like a question to ask to your SIP phones vendor. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 08/06/2010 ? 10:37, Fabio Pietrosanti (naif) a ?crit : > Hi all, > > i need to setup a server that handle VoIP connection in a way that > always works in the most efficient way respect to latency and NAT > traversal. > > FS provide very good NAT detection system but unfortunately when it > fail, it just fail and does not have any kind of fallback mechanism. > If FS do bypass-media and the client cannot communicate because the > phone call is lost. > If FS do proxy-media, the clients can always communicate (over UDP) > but > there is added latency. > > So we cannot rely only on FS NAT detection system and need also some > client side system. > > Now, in theory the best approach would be not to touch FS but just use > client-side system based on ICE/STUN with a TURN server-side component > as a fallback. > > However there's no widely available TURN server, or at least TURN is > not > widely and well diffused just now. > > I am wondering whether it may be possible to make a setup where: > - Client try to establish peer to peer connectivity with STUN/ICE > (so FS > is doing bypass-media not touching the SDP) > - If they cannot work, will go in fallback to FS (but with proxy- > media) > instead than going via TURN > > So using the FS B2BUA proxy-media feature as alternative to TURN > server, > but letting clients to dynamically try to connect directly one each > other by using the ICE methods that are dynamic. > > Does it seems something feasible? > > Fabio Pietrosanti > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/b2dd6318/attachment.html From lists at infosecurity.ch Tue Jun 8 02:00:05 2010 From: lists at infosecurity.ch (Fabio Pietrosanti (naif)) Date: Tue, 08 Jun 2010 11:00:05 +0200 Subject: [Freeswitch-users] Full NAT bypass solution with STUN/ICE/FS but without TURN: is possible? In-Reply-To: <31EED764-8AAD-4CE8-98BD-E6CE44F80162@gmail.com> References: <4C0E014A.1000507@infosecurity.ch> <31EED764-8AAD-4CE8-98BD-E6CE44F80162@gmail.com> Message-ID: <4C0E0695.1010308@infosecurity.ch> I am the SIP phone vendor, but still did not have experience with NAT bypass technologies and so i am asking whether this is a setup that may be feasible or not from the "protocols" point of view and from FS point of view. I can install a TURN server but if possible i would like to avoid setting up server-side systems other than FS. Fabio On 08/06/10 10.53, David Ponzone wrote: > Fabio, > > this sounds like a question to ask to your SIP phones vendor. > From tony.tin at noahmedia.com.hk Tue Jun 8 02:13:21 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Tue, 8 Jun 2010 17:13:21 +0800 Subject: [Freeswitch-users] LUA MySQL connection pool Message-ID: Hi, I'm using Lua script and MySQL to do dynamic dialplan. I'm wondering whether I need to create a connection pool for all calls or I just create a new connection for every call. How can I create the connection pool in Lua? Thanks. Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/fddefb5d/attachment.html From steveayre at gmail.com Tue Jun 8 02:46:45 2010 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 8 Jun 2010 10:46:45 +0100 Subject: [Freeswitch-users] Gateway Registration In-Reply-To: References: Message-ID: Configure the same gateway twice, with different names. Which gateway name you use will determine which username is used. On 7 June 2010 20:26, Samuel Macedo wrote: > Hi, > > I want to use the same Gateway but with different users and passwords to > make and receive calls. > This is the scenario: > I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch UserAgents > (Sofia External) to this gateway so I can receive inbound calls. And to make > a Outbound call I want to decide witch UserAgent I will use. > > Is there anyway to do this? > > Regards, > -- > Samuel Macedo > Belo Horizonte - Brazil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/8f68481f/attachment-0001.html From peter.olsson at visionutveckling.se Tue Jun 8 03:07:52 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 8 Jun 2010 12:07:52 +0200 Subject: [Freeswitch-users] CHANNEL_PARK event not triggered anymore from event socket? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C567CC7AECC@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C567CC7AECC@cooper> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C567CC7AFE4@cooper> I just realized I forgot to mention that it is on latest GIT, and other events show up, I'm just missing the CHANNEL_PARK event. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Olsson Skickat: den 8 juni 2010 09:09 Till: 'freeswitch-users at lists.freeswitch.org' ?mne: [Freeswitch-users] CHANNEL_PARK event not triggered anymore from event socket? I'm not filing this as a bug right now, since I don't know if this was an intended change. I used to trigger on the CHANNEL_PARK event, but it doesn't seem that this event is fired anymore. Is this change intended? If not, I will submit a jira report with full logs showing this issue. /Peter !DSPAM:4c0def1132931059720658! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/47d0de3b/attachment.html From yehavi.bourvine at gmail.com Tue Jun 8 04:44:31 2010 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Jun 2010 14:44:31 +0300 Subject: [Freeswitch-users] Multi-user on-demand conference Message-ID: Hello, Our users are used to create conference in the following way: - Call the first participant. - press "conference" which puts the other participant(s) on hold. - Call the new participant and talk to him/her. - press "conference" again to join everyone. - Repeat the above process for additional participants. They would like to do the same with Freeswitch also, and for large number of users (more than 3-4 which the phones can do themselves using the above procedure). Has anyone done this and can send an example? I would like to define one of the keys of the phone to send some * code which would behave like the above "conference" button and instructs Freeswitch to handle the conference. Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/2aa71bd3/attachment.html From freeswitch at peely.com Tue Jun 8 05:47:46 2010 From: freeswitch at peely.com (peely) Date: Tue, 8 Jun 2010 05:47:46 -0700 (PDT) Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1275862175540-5146803.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> <1275859376110-5146705.post@n2.nabble.com> <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> <1275862175540-5146803.post@n2.nabble.com> Message-ID: <1276001266045-5153498.post@n2.nabble.com> Hi again, I've checked with BT, and they are unable to support reinvites with a Require: timer. Reading the RFC, it seems that new transactions hsould not have the Require: timer but should revert to Supported: timer. Could a future build please support reinvites without the Require: timer header when sesison timers are enabled? Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5153498.html Sent from the freeswitch-users mailing list archive at Nabble.com. From oliver.schenk at iinet.net.au Tue Jun 8 06:04:14 2010 From: oliver.schenk at iinet.net.au (Oliver Schenk) Date: Tue, 08 Jun 2010 21:04:14 +0800 Subject: [Freeswitch-users] Freeswitch in an existing phone network Message-ID: <4C0E3FCE.1040803@iinet.net.au> Hi All, The company I work for currently has quite an extensive phone network which gets carried between old analogue PABXes which also has an interface to IP based phones. All the phones in our office are connected via CAT5 cable using IP, however literally hundreds of phones out in the field (we operate railway infrastructure) are on standard voice analogue phones carried through fibreoptics. Anyway, I would like to use Freeswitch purely for its IVR and TTS abilities and nothing else. So basically I just need it to act like a slave to whatever IP phone network is already out there. I'm a bit worried if I fire up freeswitch it will hijack the phone network! All our phones are accessible via a 5 digit extension. I would like Freeswitch to be behind one of those ... say 12345. If anyone within our phone network dials 12345 then Freeswitch should answer. I guess my question is...how should I go about disabling most of FreeSwitch except it's ability to pick up the phone and speak IVR/TTS and make an outgoing call via the existing phone network? Any general pointers appreciated. Thanks, Oliver Schenk From steveayre at gmail.com Tue Jun 8 06:08:50 2010 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 8 Jun 2010 14:08:50 +0100 Subject: [Freeswitch-users] Messaging: Content-Type: application/octet-stream changed to text/html. In-Reply-To: <4C0E02ED.9000204@gmx.net> References: <4C0E02ED.9000204@gmx.net> Message-ID: It looks like text/html is currently hardcoded into mod_sofia... it might be possible to adjust it though, why not open a Jira ticket as a feature request? -Steve On 8 June 2010 09:44, Peter P GMX wrote: > Hello, > > we have a SIP based Videoconferencing solution which we would like to > integrate in Freeswitch. Registering the endpoints works, however all > other operations which are done via messaging do not work as expected, > as Freeswitch changes the Content-Type, see below. > How can we tell Freeswitch not to change the Content-Type? > > Best regards > Peter > > Original message from UA to Freeswitch with "Content-Type: > application/octet-stream.": > > U 2010/06/01 18:21:57.056468 192.168.178.145:5530 -> 192.168.178.220:5060 > MESSAGE sip:835351 at my.domain SIP/2.0. > From: Hans > ;tag=de58b276076cdf119f8bcf0221b28bff. > To: sip:835351 at my.domain. > Via: SIP/2.0/UDP > 192.168.178.145:5530 > ;branch=z9hG4bKae56b276076cdf119f8bcf0221b28bff;uas-addr=192.168.178.220. > > CSeq: 3 MESSAGE. > Call-ID: 645ab276-076c-df11-824f-cf0221b28bff. > Contact: "Hans". > User-Agent: BRAVIS/0.0.0.27.4675 (Linux 2.6.32-22-generic; generic; > Ubuntu 10.04 LTS; i686; de; 8). > Content-Type: application/octet-stream. > Content-Length: 231. > . > BRVSDRABAAEAAACeAAAAGfLjfQQAAAAvrRdulaIBAAAAMQEA/wAASwUAAABzcmZseP////+n > pgi9D7MvrRduD7MBAAAANAEA/QAASwUAAABzcmZseP////+npgi9D7Onpgi9D7MBAAAAMwEA > BVHDfgQAAABob3N0/////5WmD7d9pQQUp3cPswEAAAAyAQADUcN+BAAAAGhvc3QA////laYP > t32lAgAAAA==. > > Modified message from Freeswitch to another UA with "Content-Type: > text/html": > > U 2010/06/01 18:21:57.061965 192.168.178.220:5060 -> 192.168.178.122:5530 > MESSAGE sip:835351 at 192.168.178.122:5530;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 192.168.178.220;rport;branch=z9hG4bKaHZUBrU7S2BXQ. > Max-Forwards: 70. > From: Hans > ;tag=de58b276076cdf119f8bcf0221b28bff. > To: "Peter ". > Call-ID: a3bd6fee-e83c-122d-1ca4-080027e51f59. > CSeq: 131586180 MESSAGE. > Contact:. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. > Supported: timer, precondition, path, replaces. > Content-Type: text/html. > Content-Length: 233. > . > BRVSDRABAAEAAACeAAAAGfLjfQQAAAAvrRdulaIBAAAAMQEA/wAASwUAAABzcmZseP////+n > pgi9D7MvrRduD7MBAAAANAEA/QAASwUAAABzcmZseP////+npgi9D7Onpgi9D7MBAAAAMwEA > BVHDfgQAAABob3N0/////5WmD7d9pQQUp3cPswEAAAAyAQADUcN+BAAAAGhvc3QA////laYP > t32lAgAAAA==. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/bebe8a6e/attachment.html From pjintheusa at gmail.com Tue Jun 8 06:37:54 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 8 Jun 2010 09:37:54 -0400 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: References: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> <70A84E11-A594-4534-8B98-5099A4F85EC2@gmail.com> Message-ID: Ok so I set up a simple test: and yes you appear to correct David - no SIP re-invite is involved - no RTP streams hit the box at all. nice. On Mon, Jun 7, 2010 at 5:31 PM, Code Ghar wrote: > I haven't used bypass-media extensively, either. So I am also interested in > knowing if it would introduce re-invite. I know several carriers, using some > older SIP implementations or for other reasons, do not support re-invite. In > a perfect world we would use bypass-media on the ingress and egress FS-SIP > servers, without introducing re-invite, and handle media using intermediate > FS-RTP servers. > > Aside from the media, if we use, say two FS-SIP servers, then all FS-RTP > servers can do load balance when doing egress. In this way we only need two > signaling IPs for all kinds of customers and carriers and don't have to > differentiate between ingress-only and egress-only FS-SIP servers. Each > FS-SIP is ingress and egress while all FS-RTP look only like media IPs to > all customers and carriers. Of course, I would like to get consensus > confirmation from experienced users that bypass-media does not introduce > re-invite. > > We could use a SIP Proxy, as advised by PJ, but if a bunch of FS servers > could do that same job that would be awesome, too. > > > > > On Mon, Jun 7, 2010 at 12:34 PM, David Ponzone wrote: > >> Phillip, >> >> please do :) >> >> Well, I could be wrong, but this setup should not require any re-invite. >> I never really used bypass-media on FS, but from what I understood, it >> will jut advertise the customer IP to FS-RTP-3 and FS-RTP-3's IP to the >> customer. >> >> Anyway, the idea of the design was an attempt to answer to Code's question >> at the beginning of the thread. who wanted to build a such architecture with >> FS only. >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur.* >> * >> * >> >> >> >> Le 07/06/2010 ? 19:02, Phillip Jones a ?crit : >> >> David, >> >> I hope you don't mind me interjecting here. But what is the advantage of >> your setup over the traditional SIP Proxy - FS - SIP Proxy setup? Isn't >> introducing a re-invite here, to shift the media from FS-SIP-Internal-1 to >> FS-RTP-3 introducing a further complication and potential point of failure? >> >> Pj >> >> On Mon, Jun 7, 2010 at 7:29 AM, David Ponzone wrote: >> >>> Mike, >>> >>> You're right, it can be achieved with SIP now that I think a bit more >>> about it. >>> The idea was to allow adding multiple media gateways when required, so >>> the media gateways should not be facing the carriers as some of them do >>> SIP-filtering, but should only be advertised in the SDP. >>> >>> So SIP-only boxes (doing bypass-media) should face the carriers to handle >>> the trunking. >>> In the middle, we would then have the media gateways, doing SIP and >>> mostly RTP. >>> But I guess we dont want customers to register and to send calls to a >>> media gateway, so we need another set of SIP boxes on the other side, doing >>> bypass-media also. >>> >>> So it would like this: >>> >>> ------sip-----FS-RTP-1-----sip------ >>> FS-SIP-Internal-1 >>> ------sip-----FS-RTP-2-----sip------FS-SIP-External-1----sip-----Carriers >>> ------sip-----FS-RTP-3-----sip------ >>> FS-SIP-Internal-2 >>> -------sip----FS-RTP-4-----sip------FS-SIP-External-2-----sip----Carriers >>> -------sip----FS-RTP-5-----sip------ >>> >>> Thanks to bypass-media, the RTP streams would go from customer to >>> FS-RTP-x to Carriers, and reverse. >>> And I don't see any reason why the same set of FS-SIP boxes could not be >>> used for both internal and external borders. >>> >>> Is there something wrong in this ? >>> >>> Code, does it help ? >>> >>> David Ponzone Direction Technique >>> email: david.ponzone at ipeva.fr >>> tel: 01 74 03 18 97 >>> gsm: 06 66 98 76 34 >>> >>> Service Client IPeva >>> tel: 0811 46 26 26 >>> www.ipeva.fr - www.ipeva-studio.com >>> >>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>> non autoris?e est interdite. Tout message ?lectronique est susceptible >>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >>> l'exp?diteur.* >>> * >>> * >>> >>> >>> >>> Le 05/06/2010 ? 19:54, Michael Jerris a ?crit : >>> >>> Why would it be an advantage to have your media proxies use another >>> protocol? >>> >>> Mike >>> >>> On Jun 4, 2010, at 1:59 AM, David Ponzone wrote: >>> >>> It doesn't solve the issue that all the media servers will do signaling >>> too, and will talk SIP with the carriers. >>> So the carriers will need to allow all the media servers . >>> >>> The only clean solution to avoid that, I think, is to have signaling >>> boxes allocating resources from media servers with another protocol than >>> SIP. >>> RTPproxy does that I think, but I am not sure how it works. >>> >>> David Ponzone >>> >>> _______________________________________________ >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/b6944500/attachment-0001.html From vetali100 at gmail.com Tue Jun 8 06:49:02 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Tue, 8 Jun 2010 16:49:02 +0300 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: References: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> <70A84E11-A594-4534-8B98-5099A4F85EC2@gmail.com> Message-ID: Probably you will get sip re invite only if you will use "bypass_media_* after_bridge*"* *variable*.* Vitalie 2010/6/8 Phillip Jones > Ok so I set up a simple test: > > > > > and yes you appear to correct David - no SIP re-invite is involved - no RTP > streams hit the box at all. > > nice. > > > On Mon, Jun 7, 2010 at 5:31 PM, Code Ghar wrote: > >> I haven't used bypass-media extensively, either. So I am also interested >> in knowing if it would introduce re-invite. I know several carriers, using >> some older SIP implementations or for other reasons, do not support >> re-invite. In a perfect world we would use bypass-media on the ingress and >> egress FS-SIP servers, without introducing re-invite, and handle media using >> intermediate FS-RTP servers. >> >> Aside from the media, if we use, say two FS-SIP servers, then all FS-RTP >> servers can do load balance when doing egress. In this way we only need two >> signaling IPs for all kinds of customers and carriers and don't have to >> differentiate between ingress-only and egress-only FS-SIP servers. Each >> FS-SIP is ingress and egress while all FS-RTP look only like media IPs to >> all customers and carriers. Of course, I would like to get consensus >> confirmation from experienced users that bypass-media does not introduce >> re-invite. >> >> We could use a SIP Proxy, as advised by PJ, but if a bunch of FS servers >> could do that same job that would be awesome, too. >> >> >> >> >> On Mon, Jun 7, 2010 at 12:34 PM, David Ponzone wrote: >> >>> Phillip, >>> >>> please do :) >>> >>> Well, I could be wrong, but this setup should not require any >>> re-invite. >>> I never really used bypass-media on FS, but from what I understood, it >>> will jut advertise the customer IP to FS-RTP-3 and FS-RTP-3's IP to the >>> customer. >>> >>> Anyway, the idea of the design was an attempt to answer to Code's >>> question at the beginning of the thread. who wanted to build a such >>> architecture with FS only. >>> >>> David Ponzone Direction Technique >>> email: david.ponzone at ipeva.fr >>> tel: 01 74 03 18 97 >>> gsm: 06 66 98 76 34 >>> >>> Service Client IPeva >>> tel: 0811 46 26 26 >>> www.ipeva.fr - www.ipeva-studio.com >>> >>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>> non autoris?e est interdite. Tout message ?lectronique est susceptible >>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >>> l'exp?diteur.* >>> * >>> * >>> >>> >>> >>> Le 07/06/2010 ? 19:02, Phillip Jones a ?crit : >>> >>> David, >>> >>> I hope you don't mind me interjecting here. But what is the advantage of >>> your setup over the traditional SIP Proxy - FS - SIP Proxy setup? Isn't >>> introducing a re-invite here, to shift the media from FS-SIP-Internal-1 to >>> FS-RTP-3 introducing a further complication and potential point of failure? >>> >>> Pj >>> >>> On Mon, Jun 7, 2010 at 7:29 AM, David Ponzone wrote: >>> >>>> Mike, >>>> >>>> You're right, it can be achieved with SIP now that I think a bit more >>>> about it. >>>> The idea was to allow adding multiple media gateways when required, so >>>> the media gateways should not be facing the carriers as some of them do >>>> SIP-filtering, but should only be advertised in the SDP. >>>> >>>> So SIP-only boxes (doing bypass-media) should face the carriers to >>>> handle the trunking. >>>> In the middle, we would then have the media gateways, doing SIP and >>>> mostly RTP. >>>> But I guess we dont want customers to register and to send calls to a >>>> media gateway, so we need another set of SIP boxes on the other side, doing >>>> bypass-media also. >>>> >>>> So it would like this: >>>> >>>> ------sip-----FS-RTP-1-----sip------ >>>> FS-SIP-Internal-1 >>>> ------sip-----FS-RTP-2-----sip------FS-SIP-External-1----sip-----Carriers >>>> ------sip-----FS-RTP-3-----sip------ >>>> FS-SIP-Internal-2 >>>> -------sip----FS-RTP-4-----sip------FS-SIP-External-2-----sip----Carriers >>>> -------sip----FS-RTP-5-----sip------ >>>> >>>> Thanks to bypass-media, the RTP streams would go from customer to >>>> FS-RTP-x to Carriers, and reverse. >>>> And I don't see any reason why the same set of FS-SIP boxes could not be >>>> used for both internal and external borders. >>>> >>>> Is there something wrong in this ? >>>> >>>> Code, does it help ? >>>> >>>> David Ponzone Direction Technique >>>> email: david.ponzone at ipeva.fr >>>> tel: 01 74 03 18 97 >>>> gsm: 06 66 98 76 34 >>>> >>>> Service Client IPeva >>>> tel: 0811 46 26 26 >>>> www.ipeva.fr - www.ipeva-studio.com >>>> >>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis >>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>>> non autoris?e est interdite. Tout message ?lectronique est susceptible >>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >>>> l'exp?diteur.* >>>> * >>>> * >>>> >>>> >>>> >>>> Le 05/06/2010 ? 19:54, Michael Jerris a ?crit : >>>> >>>> Why would it be an advantage to have your media proxies use another >>>> protocol? >>>> >>>> Mike >>>> >>>> On Jun 4, 2010, at 1:59 AM, David Ponzone wrote: >>>> >>>> It doesn't solve the issue that all the media servers will do signaling >>>> too, and will talk SIP with the carriers. >>>> So the carriers will need to allow all the media servers . >>>> >>>> The only clean solution to avoid that, I think, is to have signaling >>>> boxes allocating resources from media servers with another protocol than >>>> SIP. >>>> RTPproxy does that I think, but I am not sure how it works. >>>> >>>> David Ponzone >>>> >>>> _______________________________________________ >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100608/8e76eed2/attachment.html From pjintheusa at gmail.com Tue Jun 8 08:16:32 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 8 Jun 2010 11:16:32 -0400 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: References: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> <70A84E11-A594-4534-8B98-5099A4F85EC2@gmail.com> Message-ID: Yes - you are correct - I ran a test on using bypass_media_*after_bridge*and that did issue a sip-reinvite to reroute the RTP from the FS box. On Tue, Jun 8, 2010 at 9:49 AM, Vitalii Colosov wrote: > Probably you will get sip re invite only if you will use "bypass_media_* > after_bridge*"* *variable*.* > > Vitalie > > > 2010/6/8 Phillip Jones > > Ok so I set up a simple test: >> >> >> >> >> and yes you appear to correct David - no SIP re-invite is involved - no >> RTP streams hit the box at all. >> >> nice. >> >> >> On Mon, Jun 7, 2010 at 5:31 PM, Code Ghar wrote: >> >>> I haven't used bypass-media extensively, either. So I am also interested >>> in knowing if it would introduce re-invite. I know several carriers, using >>> some older SIP implementations or for other reasons, do not support >>> re-invite. In a perfect world we would use bypass-media on the ingress and >>> egress FS-SIP servers, without introducing re-invite, and handle media using >>> intermediate FS-RTP servers. >>> >>> Aside from the media, if we use, say two FS-SIP servers, then all FS-RTP >>> servers can do load balance when doing egress. In this way we only need two >>> signaling IPs for all kinds of customers and carriers and don't have to >>> differentiate between ingress-only and egress-only FS-SIP servers. Each >>> FS-SIP is ingress and egress while all FS-RTP look only like media IPs to >>> all customers and carriers. Of course, I would like to get consensus >>> confirmation from experienced users that bypass-media does not introduce >>> re-invite. >>> >>> We could use a SIP Proxy, as advised by PJ, but if a bunch of FS servers >>> could do that same job that would be awesome, too. >>> >>> >>> >>> >>> On Mon, Jun 7, 2010 at 12:34 PM, David Ponzone wrote: >>> >>>> Phillip, >>>> >>>> please do :) >>>> >>>> Well, I could be wrong, but this setup should not require any >>>> re-invite. >>>> I never really used bypass-media on FS, but from what I understood, it >>>> will jut advertise the customer IP to FS-RTP-3 and FS-RTP-3's IP to the >>>> customer. >>>> >>>> Anyway, the idea of the design was an attempt to answer to Code's >>>> question at the beginning of the thread. who wanted to build a such >>>> architecture with FS only. >>>> >>>> David Ponzone Direction Technique >>>> email: david.ponzone at ipeva.fr >>>> tel: 01 74 03 18 97 >>>> gsm: 06 66 98 76 34 >>>> >>>> Service Client IPeva >>>> tel: 0811 46 26 26 >>>> www.ipeva.fr - www.ipeva-studio.com >>>> >>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis >>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>>> non autoris?e est interdite. Tout message ?lectronique est susceptible >>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >>>> l'exp?diteur.* >>>> * >>>> * >>>> >>>> >>>> >>>> Le 07/06/2010 ? 19:02, Phillip Jones a ?crit : >>>> >>>> David, >>>> >>>> I hope you don't mind me interjecting here. But what is the advantage of >>>> your setup over the traditional SIP Proxy - FS - SIP Proxy setup? Isn't >>>> introducing a re-invite here, to shift the media from FS-SIP-Internal-1 to >>>> FS-RTP-3 introducing a further complication and potential point of failure? >>>> >>>> Pj >>>> >>>> On Mon, Jun 7, 2010 at 7:29 AM, David Ponzone wrote: >>>> >>>>> Mike, >>>>> >>>>> You're right, it can be achieved with SIP now that I think a bit more >>>>> about it. >>>>> The idea was to allow adding multiple media gateways when required, so >>>>> the media gateways should not be facing the carriers as some of them do >>>>> SIP-filtering, but should only be advertised in the SDP. >>>>> >>>>> So SIP-only boxes (doing bypass-media) should face the carriers to >>>>> handle the trunking. >>>>> In the middle, we would then have the media gateways, doing SIP and >>>>> mostly RTP. >>>>> But I guess we dont want customers to register and to send calls to a >>>>> media gateway, so we need another set of SIP boxes on the other side, doing >>>>> bypass-media also. >>>>> >>>>> So it would like this: >>>>> >>>>> ------sip-----FS-RTP-1-----sip------ >>>>> FS-SIP-Internal-1 >>>>> ------sip-----FS-RTP-2-----sip------FS-SIP-External-1----sip-----Carriers >>>>> ------sip-----FS-RTP-3-----sip------ >>>>> FS-SIP-Internal-2 >>>>> -------sip----FS-RTP-4-----sip------FS-SIP-External-2-----sip----Carriers >>>>> -------sip----FS-RTP-5-----sip------ >>>>> >>>>> Thanks to bypass-media, the RTP streams would go from customer to >>>>> FS-RTP-x to Carriers, and reverse. >>>>> And I don't see any reason why the same set of FS-SIP boxes could not >>>>> be used for both internal and external borders. >>>>> >>>>> Is there something wrong in this ? >>>>> >>>>> Code, does it help ? >>>>> >>>>> David Ponzone Direction Technique >>>>> email: david.ponzone at ipeva.fr >>>>> tel: 01 74 03 18 97 >>>>> gsm: 06 66 98 76 34 >>>>> >>>>> Service Client IPeva >>>>> tel: 0811 46 26 26 >>>>> www.ipeva.fr - www.ipeva-studio.com >>>>> >>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis >>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible >>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >>>>> l'exp?diteur.* >>>>> * >>>>> * >>>>> >>>>> >>>>> >>>>> Le 05/06/2010 ? 19:54, Michael Jerris a ?crit : >>>>> >>>>> Why would it be an advantage to have your media proxies use another >>>>> protocol? >>>>> >>>>> Mike >>>>> >>>>> On Jun 4, 2010, at 1:59 AM, David Ponzone wrote: >>>>> >>>>> It doesn't solve the issue that all the media servers will do signaling >>>>> too, and will talk SIP with the carriers. >>>>> So the carriers will need to allow all the media servers . >>>>> >>>>> The only clean solution to avoid that, I think, is to have signaling >>>>> boxes allocating resources from media servers with another protocol than >>>>> SIP. >>>>> RTPproxy does that I think, but I am not sure how it works. >>>>> >>>>> David Ponzone >>>>> >>>>> _______________________________________________ >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.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 >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/71e1fe6a/attachment-0001.html From dswardstrom at remotelink.com Tue Jun 8 09:02:36 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Tue, 8 Jun 2010 11:02:36 -0500 (CDT) Subject: [Freeswitch-users] Multi-user on-demand conference In-Reply-To: <104722469.89.1276012939728.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <1430916950.91.1276012956633.JavaMail.root@srvr12.remotelinkml.com> The JavaScript examples include a sample similar to this. With a little bit of work, this should be possible to develop. Look at http://wiki.freeswitch.org/wiki/Javascript_Examples One difference (should not be significant) is that as you add each participant, they would be added to the conference. If you use the wait-mod flag in the conference, the participants would hear music-on-hold. When the originator has added all of the participants, then add the originator as the Moderator. This action adds the originator/moderator to the conference, the MOH ends, and everyone can talk to each other. Regards,David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom From macedoslm at gmail.com Tue Jun 8 09:03:22 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Tue, 8 Jun 2010 13:03:22 -0300 Subject: [Freeswitch-users] Gateway Registration In-Reply-To: References: Message-ID: But the name of the gateway is the domain that will be used to register, or no? -- Samuel Macedo On 8 June 2010 06:46, Steven Ayre wrote: > Configure the same gateway twice, with different names. Which gateway name > you use will determine which username is used. > > > > On 7 June 2010 20:26, Samuel Macedo wrote: > >> Hi, >> >> I want to use the same Gateway but with different users and passwords to >> make and receive calls. >> This is the scenario: >> I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch >> UserAgents (Sofia External) to this gateway so I can receive inbound calls. >> And to make a Outbound call I want to decide witch UserAgent I will use. >> >> Is there anyway to do this? >> >> Regards, >> -- >> Samuel Macedo >> Belo Horizonte - Brazil >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100608/ab5c8333/attachment.html From mgende at gendesign.com Tue Jun 8 09:32:57 2010 From: mgende at gendesign.com (Michael Gende) Date: Tue, 8 Jun 2010 11:32:57 -0500 Subject: [Freeswitch-users] Out-Going Call Transfer Question Message-ID: Please forgive, I sent this follow-up a few days ago but think I accidentally deleted the response. Again, I've a PFsense FS install that works great. But, and this is probably something I did, I can't transfer a call session originating here to anywhere but a conference room (i.e., I can't transfer a call I made to another extension. It hangs up the call). For example, I just called to my cell phone from FS. Then, having answered it, transfer the call to my colleague at ext 1012. I'm extension 1013. The transfer hangs up the call on both ends. Console output follows: *2010-6-03 12:51:42:950492 [ERR] Switch_ivr_originate.c: 1494 Cannot Create outgoing channel of type [user] Cause: [SUBSCRIBER ABSENT] 2010-6-03 12:51:42:950492 [INFO] mode_dtools.c: 2091 Originate Failed. Cause: SUBSCRIBER_ABSENT* * 2010-06-03 12:51:43.960389 [WARNING] mod_voicemail.c:2931 Can't Find User [1012 at wan.ip.address (should be LAN ip typically, no?)] * > Also, is this a plain FS install with the default dialplan? If not then > include your relevant dialplan entries. Our install is pretty well documented in the following: http://wiki.freeswitch.org/wiki/Multi_home_tutorial The setup described here is what we have. > Lastly, are you on the most recent hit HEAD version of FS? > Nope. This is FS on PFSense, the Beta 0.9.7.26 version. No other issues, however, despite the Beta moniker. Is the older version the issue? Sorry for the re-post. Any comments most appreciated. Mike G. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/8e1fd050/attachment.html From jcasale at activenetwerx.com Tue Jun 8 09:46:17 2010 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Tue, 8 Jun 2010 16:46:17 +0000 Subject: [Freeswitch-users] Freeswitch in an existing phone network In-Reply-To: <4C0E3FCE.1040803@iinet.net.au> References: <4C0E3FCE.1040803@iinet.net.au> Message-ID: > I guess my >question is...how should I go about disabling most of FreeSwitch except >it's ability to pick up the phone and speak IVR/TTS and make an outgoing >call via the existing phone network? Look in conf/dialplan Rip out what's not needed... From macedoslm at gmail.com Tue Jun 8 10:32:23 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Tue, 8 Jun 2010 14:32:23 -0300 Subject: [Freeswitch-users] javascript bridge In-Reply-To: References: Message-ID: Have you had any success with this bridge code? Because I've had the same problem and my solution was: callee_dialstring = "{ignore_early_media=true,leg=2}sofia/user/XXX"; session.execute('bridge', callee_dialstring); The problem here is that you have to establish the first leg before you can dial the second number. Regards, -- Samuel Macedo On 7 June 2010 18:58, Michael Collins wrote: > I take it that this js is being called from the dialplan? If so then do > what you need to do with outsession before you bridge it. Or consider using > ESL and making a socket-based control program where you have much more > flexibility... > > -MC > > On Sun, Jun 6, 2010 at 11:55 AM, stephen at stephenjc < > stephen at stephenjc.com> wrote: > >> I have the following code, and after the bridge the javascript seems to >> stop. >> >> outsession = new >> Session("{ringback=\'%(2000,4000,440.0,480.0)\',instant_ringback=true,ignore_early_media=false}sofia/gateway/" >> + providerhash["providername0"] + "/XXXXXXXXXXXX"); >> bridge(session,outsession); >> while(outsession.ready()) >> { >> console_log("notice","ping"); >> } >> >> I am looking to manage the b leg, should i use execute_on_answer instead? >> or is there a way to make the code continue after the bridge. >> >> Thanks, >> Stephen C >> -All of my email addresses go to the same place >> -Save Paper, think before you print >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100608/29cacba5/attachment.html From delorenzodesign at gmail.com Tue Jun 8 13:08:44 2010 From: delorenzodesign at gmail.com (Michael De Lorenzo) Date: Tue, 8 Jun 2010 16:08:44 -0400 Subject: [Freeswitch-users] Call Forwarding/Diversion Dialplan Message-ID: I'm trying to write a dialplan that will forward a call to a PSTN number. PSTN (Party A) Inbound call to your Freeswitch (Party B), which is forwarded out to the PSTN (Party C) I've found examples for transferring calls to other Freeswitch users, but I haven't been able to get this to work properly--where it sets a Diversion header. Can anyone help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/617ac64e/attachment.html From sean at obscuradigital.com Tue Jun 8 14:11:08 2010 From: sean at obscuradigital.com (Sean Holt) Date: Tue, 08 Jun 2010 14:11:08 -0700 Subject: [Freeswitch-users] Early media configuration Message-ID: Hello lists, I?m having 2 issues and wondering if someone can help. Here are my specs Most recent git from freeswitch Centos 5 I?ve include my dialplan http://pastebin.freeswitch.org/13156 1st: When calling any endpoint that requires digit input, the endpoint does not recognize the dtmf. I?ve done a wireshark trace and I see the dtmf output, so I know that part is working. Tech support at Bandwidth.com says I need to enable early media, so I added the ignore_early_media=true in my dialplan but still no success. I?m thinking I don?t have ignore early media in the right place, but I?ve included my dialplan for viewing. 2nd: Incoming calls have no audio after the call bridges. Hard to say what the issue is, but it might have something to do with early media. Thoughts? Thanks Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/3fe0e690/attachment-0001.html From anthony.minessale at gmail.com Tue Jun 8 14:33:20 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Jun 2010 16:33:20 -0500 Subject: [Freeswitch-users] Need to stop more than one voice file using break application In-Reply-To: References: Message-ID: what is a signal handler and what is a normal place. based on your original request, which i did understand, I changed the code for you so you must update it and retest. On Mon, Jun 7, 2010 at 11:45 PM, Thangappan.M wrote: > > In async mode set the event lock true. > When the playback is going, just in the program waited for the playback has > to be completed. > > While playback is going, passed the signals to this process. At that time I > need to stop the current playback. > > In the signal handler using " break all" stop the playback but it > was not stopping the playback. > > If I do the " break all" in the normal place( not the signal > handler) it is being worked fine. > > > On Mon, Jun 7, 2010 at 10:38 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> I didn't understand you. >> >> >> On Mon, Jun 7, 2010 at 6:48 AM, Thangappan.M wrote: >> >>> For normal scenario it is working fine. I have done the break application >>> in the signal handler at that time it is not working. >>> >>> Any reason? >>> >>> >>> >>> >>> On Fri, Jun 4, 2010 at 9:03 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> I added a patch [eba05c3 ] so it will work now >>>> >>>> >>>> On Fri, Jun 4, 2010 at 4:01 AM, Thangappan.M wrote: >>>> >>>>> >>>>> Dear all, >>>>> >>>>> I am in the process of developing IVR using FreeSWITCH. For that I >>>>> am being used outbound ESL in async mode. >>>>> >>>>> In my design, usually for one menu it might be more than one voice >>>>> files. So using playback_delimiter, play back all the voice file in single >>>>> instance using playback application. >>>>> >>>>> I've tried to stop the play back using "break all" API. But it >>>>> only break the only one voice file not the whole application(playback). >>>>> >>>>> Consider that I am playback four voice files. When the first voice >>>>> file is getting playback, using " break all" t stop the playback . >>>>> It only stopped the second voice and continued to playback the third voice >>>>> file and followed by fourth one. >>>>> >>>>> My need is to break the playback application which may have any >>>>> number of voice files in async mode. >>>>> >>>>> Thanks in advance. >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> Thangappan.M >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Regards, >>> Thangappan.M >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100608/c28d48ba/attachment.html From anthony.minessale at gmail.com Tue Jun 8 14:42:31 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Jun 2010 16:42:31 -0500 Subject: [Freeswitch-users] CHANNEL_PARK event not triggered anymore from event socket? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C567CC7AFE4@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C567CC7AECC@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C567CC7AFE4@cooper> Message-ID: update again, a previous patch put the names for 2 new events in the wrong place throwing all the events naming off. On Tue, Jun 8, 2010 at 5:07 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > I just realized I forgot to mention that it is on latest GIT, and other > events show up, I?m just missing the CHANNEL_PARK event. > > > > /Peter > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Peter Olsson > *Skickat:* den 8 juni 2010 09:09 > *Till:* 'freeswitch-users at lists.freeswitch.org' > *?mne:* [Freeswitch-users] CHANNEL_PARK event not triggered anymore from > event socket? > > > > I?m not filing this as a bug right now, since I don?t know if this was an > intended change. I used to trigger on the CHANNEL_PARK event, but it doesn?t > seem that this event is fired anymore. Is this change intended? If not, I > will submit a jira report with full logs showing this issue. > > > > /Peter > > > > !DSPAM:4c0def1132931059720658! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100608/ea988501/attachment.html From kenfulmer at icstechnologysolutions.com Tue Jun 8 15:30:37 2010 From: kenfulmer at icstechnologysolutions.com (Ken Fulmer) Date: Tue, 8 Jun 2010 17:30:37 -0500 Subject: [Freeswitch-users] Remote Party ID Message-ID: <014d01cb075a$37d55510$a77fff30$@com> I've seen the following link but I'm still confused. http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type I realize this refers to channel variables, but is there a sip profile parameter to remove Remote Party ID? Thanks, Ken -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/47323fa6/attachment-0001.html From djbinter at gmail.com Tue Jun 8 16:24:19 2010 From: djbinter at gmail.com (DJB International) Date: Tue, 8 Jun 2010 16:24:19 -0700 Subject: [Freeswitch-users] Remote Party ID In-Reply-To: <014d01cb075a$37d55510$a77fff30$@com> References: <014d01cb075a$37d55510$a77fff30$@com> Message-ID: On Tue, Jun 8, 2010 at 3:30 PM, Ken Fulmer < kenfulmer at icstechnologysolutions.com> wrote: > I?ve seen the following link but I?m still confused. > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type > > > > I realize this refers to channel variables, but is there a sip profile > parameter to remove Remote Party ID? > > > > Thanks, > > > > Ken > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/5a84542b/attachment.html From djbinter at gmail.com Tue Jun 8 16:37:54 2010 From: djbinter at gmail.com (DJB International) Date: Tue, 8 Jun 2010 16:37:54 -0700 Subject: [Freeswitch-users] Remote Party ID In-Reply-To: References: <014d01cb075a$37d55510$a77fff30$@com> Message-ID: I also updated the wiki. You can find it here: http://wiki.freeswitch.org/wiki/Sofia.conf.xml -djbinter On Tue, Jun 8, 2010 at 4:24 PM, DJB International wrote: > > > > > > > > > > > > > On Tue, Jun 8, 2010 at 3:30 PM, Ken Fulmer < > kenfulmer at icstechnologysolutions.com> wrote: > >> I?ve seen the following link but I?m still confused. >> >> >> >> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type >> >> >> >> I realize this refers to channel variables, but is there a sip profile >> parameter to remove Remote Party ID? >> >> >> >> Thanks, >> >> >> >> Ken >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/7de393d2/attachment.html From robert.hadley at teotech.com Tue Jun 8 16:44:43 2010 From: robert.hadley at teotech.com (Robert Hadley) Date: Tue, 8 Jun 2010 16:44:43 -0700 Subject: [Freeswitch-users] application=record_session leaves a "session" around? Message-ID: Using seems to leave a "session" around. I notice this repeated message later when I restart sofia internal profile after recording calls: 2010-06-08 15:37:56.641151 [CRIT] sofia.c:1459 Waiting for 3 session(s) (Msg repeats 13 times) The number of sessions "Waiting for" is the number of calls I have recorded. Pastebin of console debug log at: http://pastebin.freeswitch.org/13157 Is this a bug or is there something I need to do after recording a call to free the session? Thanks, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/568b614c/attachment.html From brian at freeswitch.org Tue Jun 8 16:58:05 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Jun 2010 18:58:05 -0500 Subject: [Freeswitch-users] application=record_session leaves a "session" around? In-Reply-To: References: Message-ID: Have you made sure you're on the very latest FreeSWITCH code? /b On Jun 8, 2010, at 6:44 PM, Robert Hadley wrote: > Using seems to leave a ?session? around. > I notice this repeated message later when I restart sofia internal profile after recording calls: > 2010-06-08 15:37:56.641151 [CRIT] sofia.c:1459 Waiting for 3 session(s) (Msg repeats 13 times) > The number of sessions ?Waiting for? is the number of calls I have recorded. > Pastebin of console debug log at: http://pastebin.freeswitch.org/13157 > > Is this a bug or is there something I need to do after recording a call to free the session? > > Thanks, > Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/d3572edb/attachment.html From engineerzuhairraza at gmail.com Tue Jun 8 21:48:14 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Wed, 9 Jun 2010 09:48:14 +0500 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: Thank you sir it clearified everything. I have one more question to ask, I am unable to compile alsa-driver-1.0.20 on centos 5.5, I googled but could not find any usable info I got this error. In file included from /usr/src/alsa-driver-1.0.20/acore/hwdep.c:1: /usr/src/alsa-driver-1.0.20/include/adriver.h:1779:1: error: unterminated #if /usr/src/alsa-driver-1.0.20/include/adriver.h:1:1: error: unterminated #ifndef make[3]: *** [/usr/src/alsa-driver-1.0.20/acore/hwdep.o] Error 1 make[2]: *** [/usr/src/alsa-driver-1.0.20/acore] Error 2 make[1]: *** [_module_/usr/src/alsa-driver-1.0.20] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-194.3.1.el5-i686' make: *** [compile] Error 2 Then I tried to compile alsar-drive-1.0.21 it was successfully compiled and installed but my skype client does not show dummy hardware. What should i do? On Mon, Jun 7, 2010 at 5:19 PM, Giovanni Maruzzelli wrote: > Zuhair, > > maybe you need to find a friend with some more unix experience, that > can help you for the first few days. > > You have to: > > 1) compile skypopen_auth as in the wiki ("gcc -Wall -ggdb > skypopen_auth.c -o skypopen_auth -lX11") > 2) login with ssh -X > 3) launch Skype client > 4) launch "skypopen_auth localhost:10.0" > > -giovanni > > On Mon, Jun 7, 2010 at 2:05 PM, Zuhair Raza > wrote: > > you mean to use this syntax > > gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:101 ?? > > > > When i do echo $DISPLAY it says localhost:10.0 > > and when gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:$DISPLAY > > it says cannot find -lX11:localhost:10.0 > > > > > > On Mon, Jun 7, 2010 at 4:36 PM, Giovanni Maruzzelli > > > wrote: > >> > >> On Mon, Jun 7, 2010 at 1:29 PM, Zuhair Raza > >> wrote: > >> > ok thanks.. but what about my question that skype didnot ask me to to > >> > connect to skype API. and when i do either on ./skypopen_auth :101 or > >> > :0 it > >> > says cannot open X display > >> > >> I told you in previous mail. > >> > >> You need to launch skypopen_auth from the same ssh -X from which you > >> launched the Skype client. And you must give skypopen_auth the correct > >> xserver as an argument. > >> You can use $DISPLAY, or you can check it with "echo $DISPLAY" and > >> then use that value. > >> > >> > >> > > >> > On Mon, Jun 7, 2010 at 3:53 PM, Giovanni Maruzzelli > >> > > >> > wrote: > >> >> > >> >> On Mon, Jun 7, 2010 at 12:45 PM, Zuhair Raza > >> >> wrote: > >> >> > Thanks for explanation Sir, One more question, please explain > >> >> > > >> >> > cd /root > >> >> > mount /dev/hda5 /mnt > >> >> > cp /mnt/root/skypeconfig2.tgz ./ > >> >> > tar xzf skypeconfig2.tgz > >> >> > chown -R root.root .Skype > >> >> > > >> >> > According to wiki we haven't created a tgz file before that, but > >> >> > .Skype > >> >> > directory at the server > >> >> > >> >> If you do it with ssh -X (as pre the previous mail), you don't need a > >> >> Skype config directory from another computer. You created that > >> >> directory. > >> >> So, just skip those steps. > >> >> > >> >> -giovanni > >> >> > >> >> > > >> >> > > >> >> > On Mon, Jun 7, 2010 at 12:07 PM, Giovanni Maruzzelli > >> >> > > >> >> > wrote: > >> >> >> > >> >> >> On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza > >> >> >> wrote: > >> >> >> >> version. First i enter command at my freeswitch box under > >> >> >> >> mod_skypopen/configs > >> >> >> >> > >> >> >> >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 > >> >> >> >> > >> >> >> >> then I install xauth on server after that i ssh with x > forwarding > >> >> >> >> to > >> >> >> >> my > >> >> >> >> server from an another linux desktop and open skype by typing > >> >> >> >> /usr/bin/skype, it launched skype client at Linux desktop but > it > >> >> >> >> didn't > >> >> >> >> ask > >> >> >> >> for connecting with skypopen api, although it creates ".Skype" > >> >> >> >> directory on > >> >> >> > >> >> >> After launching the skype client from the ssh -X session, you have > >> >> >> to > >> >> >> launch skypopen_auth from the same ssh -X session (giving the X > >> >> >> server > >> >> >> as an argument), eg: "./skypopen_auth $DISPLAY" > >> >> >> > >> >> >> >> my server but when i load mod_skypopen it says could not find > any > >> >> >> >> skype > >> >> >> >> instance and when i typed skypopen_auth it also says no skype > >> >> >> >> instance > >> >> >> >> found > >> >> >> >> on X 0:0. Can anyone tell me where I have mistaken?? > >> >> >> > >> >> >> After having given the auth to the skype client, and closing it so > >> >> >> it > >> >> >> save that auth, you close the ssh -X session, and launch an X > server > >> >> >> and a skype client in the server, using the script in the configs > >> >> >> directory (as explained in the wiki). > >> >> >> > >> >> >> After having launched that script, you load mod_skypopen. > >> >> >> > >> >> >> You MUST edit the script and the skypopen.conf.xml to use your own > >> >> >> values for skype username and password. > >> >> >> > >> >> >> -- > >> >> >> Sincerely, > >> >> >> > >> >> >> Giovanni Maruzzelli > >> >> >> Cell : +39-347-2665618 > >> >> >> > >> >> >> _______________________________________________ > >> >> >> FreeSWITCH-users mailing list > >> >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> > >> >> >> > >> >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >> http://www.freeswitch.org > >> >> > > >> >> > > >> >> > > >> >> > -- > >> >> > Regards, > >> >> > Zuhair Raza > >> >> > > >> >> > > >> >> > _______________________________________________ > >> >> > FreeSWITCH-users mailing list > >> >> > FreeSWITCH-users at 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 > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > > >> > -- > >> > Regards, > >> > Zuhair Raza > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Regards, > > Zuhair Raza > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Zuhair Raza -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/2ad259a0/attachment-0001.html From babak.freeswitch at gmail.com Tue Jun 8 22:06:26 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Wed, 9 Jun 2010 09:36:26 +0430 Subject: [Freeswitch-users] odbc on windows In-Reply-To: References: Message-ID: so thanx in advance :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/d3d2cbfc/attachment.html From infos at madovsky.org Tue Jun 8 23:08:48 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 9 Jun 2010 02:08:48 -0400 Subject: [Freeswitch-users] mod_lcr and memcache Message-ID: Is mod_lcr uses memcache for DB request ? Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/85494120/attachment.html From peter.olsson at visionutveckling.se Wed Jun 9 01:21:18 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 9 Jun 2010 10:21:18 +0200 Subject: [Freeswitch-users] CHANNEL_PARK event not triggered anymore from event socket? In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C567CC7AECC@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C567CC7AFE4@cooper> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C567CC7B371@cooper> Thanks Tony - as always, you fixed the problem :) /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 8 juni 2010 23:43 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] CHANNEL_PARK event not triggered anymore from event socket? update again, a previous patch put the names for 2 new events in the wrong place throwing all the events naming off. On Tue, Jun 8, 2010 at 5:07 AM, Peter Olsson > wrote: I just realized I forgot to mention that it is on latest GIT, and other events show up, I'm just missing the CHANNEL_PARK event. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Olsson Skickat: den 8 juni 2010 09:09 Till: 'freeswitch-users at lists.freeswitch.org' ?mne: [Freeswitch-users] CHANNEL_PARK event not triggered anymore from event socket? I'm not filing this as a bug right now, since I don't know if this was an intended change. I used to trigger on the CHANNEL_PARK event, but it doesn't seem that this event is fired anymore. Is this change intended? If not, I will submit a jira report with full logs showing this issue. /Peter _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 !DSPAM:4c0ebb0f32932533525830! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/49d4d09a/attachment.html From gmaruzz at celliax.org Wed Jun 9 04:59:55 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 9 Jun 2010 13:59:55 +0200 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: On Wed, Jun 9, 2010 at 6:48 AM, Zuhair Raza wrote: > I have one more question to ask, I am unable to compile alsa-driver-1.0.20 > on centos 5.5, I googled but could not find any usable info I got this > error. > > In file included from /usr/src/alsa-driver-1.0.20/acore/hwdep.c:1: > /usr/src/alsa-driver-1.0.20/include/adriver.h:1779:1: error: unterminated > #if > /usr/src/alsa-driver-1.0.20/include/adriver.h:1:1: error: unterminated > #ifndef > make[3]: *** [/usr/src/alsa-driver-1.0.20/acore/hwdep.o] Error 1 > make[2]: *** [/usr/src/alsa-driver-1.0.20/acore] Error 2 > make[1]: *** [_module_/usr/src/alsa-driver-1.0.20] Error 2 > make[1]: Leaving directory `/usr/src/kernels/2.6.18-194.3.1.el5-i686' > make: *** [compile] Error 2 > use alsa drivers 1.0.20 and compile them as per the instructions on the wiki (you have to edit some files, is very clearly explained in the wiki page). The preferred centos version for FreeSWITCH is 5.3, but 5.5 would probably be ok. Maybe with centos 5.5 they changed something, in that case you will have to do adapt the dits to the files. > > On Mon, Jun 7, 2010 at 5:19 PM, Giovanni Maruzzelli > wrote: >> >> Zuhair, >> >> maybe you need to find a friend with some more unix experience, that >> can help you for the first few days. >> >> You have to: >> >> 1) compile skypopen_auth as in the wiki ("gcc -Wall -ggdb >> skypopen_auth.c -o skypopen_auth -lX11") >> 2) login with ssh -X >> 3) launch Skype client >> 4) launch "skypopen_auth localhost:10.0" >> >> -giovanni >> >> On Mon, Jun 7, 2010 at 2:05 PM, Zuhair Raza >> wrote: >> > you mean to use this syntax >> > gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:101? ?? >> > >> > When i do echo $DISPLAY it says localhost:10.0 >> > and when gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:$DISPLAY >> > it says cannot find -lX11:localhost:10.0 >> > >> > >> > On Mon, Jun 7, 2010 at 4:36 PM, Giovanni Maruzzelli >> > >> > wrote: >> >> >> >> On Mon, Jun 7, 2010 at 1:29 PM, Zuhair Raza >> >> wrote: >> >> > ok thanks.. but what about my question that skype didnot ask me to to >> >> > connect to skype API.? and when i do either on ./skypopen_auth :101 >> >> > or >> >> > :0 it >> >> > says cannot open X display >> >> >> >> I told you in previous mail. >> >> >> >> You need to launch skypopen_auth from the same ssh -X from which you >> >> launched the Skype client. And you must give skypopen_auth the correct >> >> xserver as an argument. >> >> You can use $DISPLAY, or you can check it with "echo $DISPLAY" and >> >> then use that value. >> >> >> >> >> >> > >> >> > On Mon, Jun 7, 2010 at 3:53 PM, Giovanni Maruzzelli >> >> > >> >> > wrote: >> >> >> >> >> >> On Mon, Jun 7, 2010 at 12:45 PM, Zuhair Raza >> >> >> wrote: >> >> >> > Thanks for explanation Sir, One more question, please explain >> >> >> > >> >> >> > cd /root >> >> >> > mount /dev/hda5 /mnt >> >> >> > cp /mnt/root/skypeconfig2.tgz ./ >> >> >> > tar xzf skypeconfig2.tgz >> >> >> > chown -R root.root .Skype >> >> >> > >> >> >> > According to wiki we haven't created a tgz file before that, but >> >> >> > .Skype >> >> >> > directory at the server >> >> >> >> >> >> If you do it with ssh -X (as pre the previous mail), you don't need >> >> >> a >> >> >> Skype config directory from another computer. You created that >> >> >> directory. >> >> >> So, just skip those steps. >> >> >> >> >> >> -giovanni >> >> >> >> >> >> > >> >> >> > >> >> >> > On Mon, Jun 7, 2010 at 12:07 PM, Giovanni Maruzzelli >> >> >> > >> >> >> > wrote: >> >> >> >> >> >> >> >> On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza >> >> >> >> wrote: >> >> >> >> >> version. First i enter command at my freeswitch box under >> >> >> >> >> mod_skypopen/configs >> >> >> >> >> >> >> >> >> >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 >> >> >> >> >> >> >> >> >> >> then I install xauth on server after that i ssh with x >> >> >> >> >> forwarding >> >> >> >> >> to >> >> >> >> >> my >> >> >> >> >> server from an another linux desktop and open skype by typing >> >> >> >> >> /usr/bin/skype, it launched skype client at Linux desktop but >> >> >> >> >> it >> >> >> >> >> didn't >> >> >> >> >> ask >> >> >> >> >> for connecting with skypopen api, although it creates ".Skype" >> >> >> >> >> directory on >> >> >> >> >> >> >> >> After launching the skype client from the ssh -X session, you >> >> >> >> have >> >> >> >> to >> >> >> >> launch skypopen_auth from the same ssh -X session (giving the X >> >> >> >> server >> >> >> >> as an argument), eg: "./skypopen_auth $DISPLAY" >> >> >> >> >> >> >> >> >> my server but when i load mod_skypopen it says could not find >> >> >> >> >> any >> >> >> >> >> skype >> >> >> >> >> instance and when i typed skypopen_auth it also says no skype >> >> >> >> >> instance >> >> >> >> >> found >> >> >> >> >> on X 0:0. Can anyone tell me where I have mistaken?? >> >> >> >> >> >> >> >> After having given the auth to the skype client, and closing it >> >> >> >> so >> >> >> >> it >> >> >> >> save that auth, you close the ssh -X session, and launch an X >> >> >> >> server >> >> >> >> and a skype client in the server, using the script in the configs >> >> >> >> directory (as explained in the wiki). >> >> >> >> >> >> >> >> After having launched that script, you load mod_skypopen. >> >> >> >> >> >> >> >> You MUST edit the script and the skypopen.conf.xml to use your >> >> >> >> own >> >> >> >> values for skype username and password. >> >> >> >> >> >> >> >> -- >> >> >> >> Sincerely, >> >> >> >> >> >> >> >> Giovanni Maruzzelli >> >> >> >> Cell : +39-347-2665618 >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> FreeSWITCH-users mailing list >> >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >> http://www.freeswitch.org >> >> >> > >> >> >> > >> >> >> > >> >> >> > -- >> >> >> > Regards, >> >> >> > Zuhair Raza >> >> >> > >> >> >> > >> >> >> > _______________________________________________ >> >> >> > FreeSWITCH-users mailing list >> >> >> > FreeSWITCH-users at 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 >> >> >> >> >> >> _______________________________________________ >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > -- >> >> > Regards, >> >> > Zuhair Raza >> >> > >> >> > >> >> > _______________________________________________ >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at 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 >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Regards, >> > Zuhair Raza >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Regards, > Zuhair Raza > > > > > -- > Regards, > Zuhair Raza > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From gmaruzz at celliax.org Wed Jun 9 05:03:41 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 9 Jun 2010 14:03:41 +0200 Subject: [Freeswitch-users] Hi In-Reply-To: References: Message-ID: also, if you use alsa 1.0.21 you just follow the same instructions in the wiki, and maybe the edits will not be necessary. The custom snd-dummy is ok for all alsa versions But you will have to modprobe it, I mean you have to load the dummy driver. From the command line, as root: modprobe snd-dummy or just use the script for launching the Skype client. I repeat: maybe will be better if you ask some friends that knows unix-linux to help you in the first days, or you will have many quirks like this one. ciao, -giovanni On Wed, Jun 9, 2010 at 1:59 PM, Giovanni Maruzzelli wrote: > On Wed, Jun 9, 2010 at 6:48 AM, Zuhair Raza > wrote: >> I have one more question to ask, I am unable to compile alsa-driver-1.0.20 >> on centos 5.5, I googled but could not find any usable info I got this >> error. >> >> In file included from /usr/src/alsa-driver-1.0.20/acore/hwdep.c:1: >> /usr/src/alsa-driver-1.0.20/include/adriver.h:1779:1: error: unterminated >> #if >> /usr/src/alsa-driver-1.0.20/include/adriver.h:1:1: error: unterminated >> #ifndef >> make[3]: *** [/usr/src/alsa-driver-1.0.20/acore/hwdep.o] Error 1 >> make[2]: *** [/usr/src/alsa-driver-1.0.20/acore] Error 2 >> make[1]: *** [_module_/usr/src/alsa-driver-1.0.20] Error 2 >> make[1]: Leaving directory `/usr/src/kernels/2.6.18-194.3.1.el5-i686' >> make: *** [compile] Error 2 >> > > use alsa drivers 1.0.20 and compile them as per the instructions on > the wiki (you have to edit some files, is very clearly explained in > the wiki page). > The preferred centos version for FreeSWITCH is 5.3, but 5.5 would > probably be ok. > Maybe with centos 5.5 they changed something, in that case you will > have to do adapt the dits to the files. > >> >> On Mon, Jun 7, 2010 at 5:19 PM, Giovanni Maruzzelli >> wrote: >>> >>> Zuhair, >>> >>> maybe you need to find a friend with some more unix experience, that >>> can help you for the first few days. >>> >>> You have to: >>> >>> 1) compile skypopen_auth as in the wiki ("gcc -Wall -ggdb >>> skypopen_auth.c -o skypopen_auth -lX11") >>> 2) login with ssh -X >>> 3) launch Skype client >>> 4) launch "skypopen_auth localhost:10.0" >>> >>> -giovanni >>> >>> On Mon, Jun 7, 2010 at 2:05 PM, Zuhair Raza >>> wrote: >>> > you mean to use this syntax >>> > gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:101? ?? >>> > >>> > When i do echo $DISPLAY it says localhost:10.0 >>> > and when gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:$DISPLAY >>> > it says cannot find -lX11:localhost:10.0 >>> > >>> > >>> > On Mon, Jun 7, 2010 at 4:36 PM, Giovanni Maruzzelli >>> > >>> > wrote: >>> >> >>> >> On Mon, Jun 7, 2010 at 1:29 PM, Zuhair Raza >>> >> wrote: >>> >> > ok thanks.. but what about my question that skype didnot ask me to to >>> >> > connect to skype API.? and when i do either on ./skypopen_auth :101 >>> >> > or >>> >> > :0 it >>> >> > says cannot open X display >>> >> >>> >> I told you in previous mail. >>> >> >>> >> You need to launch skypopen_auth from the same ssh -X from which you >>> >> launched the Skype client. And you must give skypopen_auth the correct >>> >> xserver as an argument. >>> >> You can use $DISPLAY, or you can check it with "echo $DISPLAY" and >>> >> then use that value. >>> >> >>> >> >>> >> > >>> >> > On Mon, Jun 7, 2010 at 3:53 PM, Giovanni Maruzzelli >>> >> > >>> >> > wrote: >>> >> >> >>> >> >> On Mon, Jun 7, 2010 at 12:45 PM, Zuhair Raza >>> >> >> wrote: >>> >> >> > Thanks for explanation Sir, One more question, please explain >>> >> >> > >>> >> >> > cd /root >>> >> >> > mount /dev/hda5 /mnt >>> >> >> > cp /mnt/root/skypeconfig2.tgz ./ >>> >> >> > tar xzf skypeconfig2.tgz >>> >> >> > chown -R root.root .Skype >>> >> >> > >>> >> >> > According to wiki we haven't created a tgz file before that, but >>> >> >> > .Skype >>> >> >> > directory at the server >>> >> >> >>> >> >> If you do it with ssh -X (as pre the previous mail), you don't need >>> >> >> a >>> >> >> Skype config directory from another computer. You created that >>> >> >> directory. >>> >> >> So, just skip those steps. >>> >> >> >>> >> >> -giovanni >>> >> >> >>> >> >> > >>> >> >> > >>> >> >> > On Mon, Jun 7, 2010 at 12:07 PM, Giovanni Maruzzelli >>> >> >> > >>> >> >> > wrote: >>> >> >> >> >>> >> >> >> On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza >>> >> >> >> wrote: >>> >> >> >> >> version. First i enter command at my freeswitch box under >>> >> >> >> >> mod_skypopen/configs >>> >> >> >> >> >>> >> >> >> >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 >>> >> >> >> >> >>> >> >> >> >> then I install xauth on server after that i ssh with x >>> >> >> >> >> forwarding >>> >> >> >> >> to >>> >> >> >> >> my >>> >> >> >> >> server from an another linux desktop and open skype by typing >>> >> >> >> >> /usr/bin/skype, it launched skype client at Linux desktop but >>> >> >> >> >> it >>> >> >> >> >> didn't >>> >> >> >> >> ask >>> >> >> >> >> for connecting with skypopen api, although it creates ".Skype" >>> >> >> >> >> directory on >>> >> >> >> >>> >> >> >> After launching the skype client from the ssh -X session, you >>> >> >> >> have >>> >> >> >> to >>> >> >> >> launch skypopen_auth from the same ssh -X session (giving the X >>> >> >> >> server >>> >> >> >> as an argument), eg: "./skypopen_auth $DISPLAY" >>> >> >> >> >>> >> >> >> >> my server but when i load mod_skypopen it says could not find >>> >> >> >> >> any >>> >> >> >> >> skype >>> >> >> >> >> instance and when i typed skypopen_auth it also says no skype >>> >> >> >> >> instance >>> >> >> >> >> found >>> >> >> >> >> on X 0:0. Can anyone tell me where I have mistaken?? >>> >> >> >> >>> >> >> >> After having given the auth to the skype client, and closing it >>> >> >> >> so >>> >> >> >> it >>> >> >> >> save that auth, you close the ssh -X session, and launch an X >>> >> >> >> server >>> >> >> >> and a skype client in the server, using the script in the configs >>> >> >> >> directory (as explained in the wiki). >>> >> >> >> >>> >> >> >> After having launched that script, you load mod_skypopen. >>> >> >> >> >>> >> >> >> You MUST edit the script and the skypopen.conf.xml to use your >>> >> >> >> own >>> >> >> >> values for skype username and password. >>> >> >> >> >>> >> >> >> -- >>> >> >> >> Sincerely, >>> >> >> >> >>> >> >> >> Giovanni Maruzzelli >>> >> >> >> Cell : +39-347-2665618 >>> >> >> >> >>> >> >> >> _______________________________________________ >>> >> >> >> FreeSWITCH-users mailing list >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org >>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >> >> http://www.freeswitch.org >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> > -- >>> >> >> > Regards, >>> >> >> > Zuhair Raza >>> >> >> > >>> >> >> > >>> >> >> > _______________________________________________ >>> >> >> > FreeSWITCH-users mailing list >>> >> >> > FreeSWITCH-users at 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 >>> >> >> >>> >> >> _______________________________________________ >>> >> >> FreeSWITCH-users mailing list >>> >> >> FreeSWITCH-users at lists.freeswitch.org >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >> >>> >> >> >>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >> http://www.freeswitch.org >>> >> > >>> >> > >>> >> > >>> >> > -- >>> >> > Regards, >>> >> > Zuhair Raza >>> >> > >>> >> > >>> >> > _______________________________________________ >>> >> > FreeSWITCH-users mailing list >>> >> > FreeSWITCH-users at 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 >>> >> >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > >>> > -- >>> > Regards, >>> > Zuhair Raza >>> > >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Regards, >> Zuhair Raza >> >> >> >> >> -- >> Regards, >> Zuhair Raza >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From lakindia89 at gmail.com Wed Jun 9 05:06:58 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Wed, 9 Jun 2010 17:36:58 +0530 Subject: [Freeswitch-users] Receiving SEGV on Outbound ESL Message-ID: Hi all, I'm trying to do the following scenario. I've created an Inbound socket. I issue the following command in the Inbound ESL api originate {origination_caller_id_name='lakshmanan',origination_caller_id_number=227,call=1,user=2,type=test,id=1001,refid=847,enabled=0,ignore_early_media=true,originate_timeout=30}user/1006 202 XML default My dialplan looks like follows: So once the user 1006 answers the call, it will connect to an Outbound socket. And here is my program in perl for controlling the Outbound ESL. use lib '/root/freeswitch/libs/esl/perl/'; require ESL; use IO::Socket::INET; my $ip = "localhost"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', Proto => 'tcp', Listen => 1, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; for(;;) { printf "Going to wait for clients to connect\n\n"; my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { close($new_sock); next; } ®isterSignals(); my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); printf "Before making a ESL connection $fd\n"; my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s\n", $uuid, $info->getHeader("caller-caller-id-number"); print "BYE\n"; close($new_sock); } sub registerSignals() { foreach ( keys %SIG ) { $SIG{$_} = 'sig_Handler'; } } sub sig_Handler() { my $handle = shift; print "Got signal $handle\n"; if($handle eq "CHLD") { wait(); } } But as soon as the call comes to the Outbound ESL program, it receives SIGSEGV. The child program prints the following: Before making a ESL connection 4 Got signal SEGV Can any one please tell me why this is happening?? And one more thing. If I execute the following API, it works fine as expected. api originate {origination_caller_id_name='lakshmanan',origination_caller_id_number=227,call=1}user/1006 202 XML default The child prints as follows: Before making a ESL connection 4 Connected call 01200ab6-73bf-11df-af54-55cf82f02c9b, from 227 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/0d22b8b6/attachment.html From jonpauldavies at gmail.com Wed Jun 9 05:46:57 2010 From: jonpauldavies at gmail.com (Jon Davies) Date: Wed, 9 Jun 2010 13:46:57 +0100 Subject: [Freeswitch-users] Non-Blocking Music on Hold Message-ID: Hi There I'm attempting to port an application I've written in CCXML/VXML to FreeSwitch/javascript and finding it a very enjoyable experience. I've hit the limits of the documentation and my knowledge and was wondering if there were any clever people on the list who can assist me with one little part. My scenario is: Make an outbound call and play it MOH Make a 2nd outbound call If the 2nd call connects, take the 1st call off MOH and bridge them together If the 2nd call fails for whatever reason, take the 1st call off MOH and play it a 'sorry' message. My problem is, all my attempts to play MOH to the first call results in blocking, meaning I dont ever get to originate my 2nd call. I've tried firstcall.streamFile("music.wav"); and firstcall.execute( "fifo", "myqueue in undef 'music.wav'" ); but both block until I hang up. Anyone got any insight as to how I can achieve this scenario in a non-blocking manner? I'm using javascript. -- Regards Jon Davies From jan.berger at video24.no Wed Jun 9 06:19:27 2010 From: jan.berger at video24.no (Jan Berger) Date: Wed, 9 Jun 2010 15:19:27 +0200 Subject: [Freeswitch-users] API pages on www.freeswitch.org Message-ID: <544C6ACCF81B47E3A7A3E41BD23F51E5@dell9400> Hi, I try to access the API pages on the freeswitch site, but I have so far never been able to - they just time out? Is this just me or do others have the same problem? Jan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/8a1e65ce/attachment.html From vetali100 at gmail.com Wed Jun 9 06:36:32 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Wed, 9 Jun 2010 16:36:32 +0300 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? Message-ID: Hi, I am trying to make 2 freeswitches (bridged) to work both in bypass media mode - currently with no luck. Would appreciate your advice. Let say I have 2 FreeSWITCH servers - one acts as PROXY (in bypass_media_after_bridge mode) and second processes media. I make a call from one sip client to another via these 2 servers (let's say G711 is used everywhere). SIP CLIENT1 -> FS1 (proxy) -> FS2 (media) -> SIP CLIENT2 Proxy server (FS1) sends a re-INVITE and excludes itself from the media path after the call was answered (I am using bypass_media_after_bridge). So, the media goes like this: SIP CLIENT1 <-> FS2 <-> SIP CLIENT2 Now, I want to try to exclude the media server (FS2) as well (just experimenting, but if both clients use same codec, I suppose this can be legitimately done?). I put bypass_media_after_bridge=true on the media server (I did it in few places in Lua script). *But as per sip trace, it does not sends any reinvite after the call was answered.* *Only FS1 sends reINVITE.* So, looks like I cannot remove FS2 from media path like this. Do you think it is not possible to acheive this kind of configuration so both servers will work in bypass media? Thank you, Vitalie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/6a409bc4/attachment.html From rupa at rupa.com Wed Jun 9 06:48:44 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 9 Jun 2010 08:48:44 -0500 Subject: [Freeswitch-users] mod_lcr and memcache In-Reply-To: References: Message-ID: No On Wed, Jun 9, 2010 at 1:08 AM, Madovsky wrote: > Is mod_lcr uses memcache for DB request ? > > Thanks > > Franck > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/45cc801b/attachment.html From kenfulmer at icstechnologysolutions.com Wed Jun 9 06:50:04 2010 From: kenfulmer at icstechnologysolutions.com (Ken Fulmer) Date: Wed, 9 Jun 2010 08:50:04 -0500 Subject: [Freeswitch-users] Remote Party ID In-Reply-To: References: <014d01cb075a$37d55510$a77fff30$@com> Message-ID: <005b01cb07da$aa53ba70$fefb2f50$@com> Thank you for replying and modifying the Wiki page. However, I'm not able to get this to work. I tried the P-asserted identity as well with no luck. Here's what I have configured for our connection to PaeTec: /// PAETEC /// /// account username *required* /// /// auth realm: *optional* same as gateway name, if blank /// /// username to use in from: *optional* same as username, if blank /// /// domain to use in from: *optional* same as realm, if blank /// /// account password *required* /// /// extension for inbound calls: *optional* same as username, if blank /// /// proxy host: *optional* same as realm, if blank /// /// send register to this proxy: *optional* same as proxy, if blank /// /// expire in seconds: *optional* 3600, if blank /// /// register true / false/// /// which transport to use for register /// How many seconds before a retry when a failure or timeout occurs /// Use the callerid of an inbound call in the from field on outbound calls via this gateway /// extra sip params to send in the contact /// send an options ping every x seconds, failure will unregister and/or mark it down /// Replaces the "gw + external" in the Contact field with the Extension value. /// Removes the default Remote Party ID. Thanks, Ken From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of DJB International Sent: Tuesday, June 08, 2010 6:24 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Remote Party ID On Tue, Jun 8, 2010 at 3:30 PM, Ken Fulmer wrote: I've seen the following link but I'm still confused. http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type I realize this refers to channel variables, but is there a sip profile parameter to remove Remote Party ID? Thanks, Ken _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/e814c264/attachment-0001.html From brian at freeswitch.org Wed Jun 9 07:01:16 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Jun 2010 09:01:16 -0500 Subject: [Freeswitch-users] Receiving SEGV on Outbound ESL In-Reply-To: References: Message-ID: <5583C105-20E2-49DE-BE53-696BEF45B1FD@freeswitch.org> Have you updated to the latest code in git? /b On Jun 9, 2010, at 7:06 AM, lakshmanan ganapathy wrote: > But as soon as the call comes to the Outbound ESL program, it receives SIGSEGV. > The child program prints the following: > Before making a ESL connection 4 > Got signal SEGV > > > Can any one please tell me why this is happening?? > > And one more thing. If I execute the following API, it works fine as expected. From msc at freeswitch.org Wed Jun 9 07:02:31 2010 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 9 Jun 2010 07:02:31 -0700 Subject: [Freeswitch-users] API pages on www.freeswitch.org In-Reply-To: <544C6ACCF81B47E3A7A3E41BD23F51E5@dell9400> References: <544C6ACCF81B47E3A7A3E41BD23F51E5@dell9400> Message-ID: <3ED7EAE4-C838-4634-885A-49DE83B346FB@freeswitch.org> It's just a link to docs.freeswitch.org. I just checked it and worked ok for me. -MC Sent from my iPhone On Jun 9, 2010, at 6:19 AM, "Jan Berger" wrote: > Hi, > > > > I try to access the API pages on the freeswitch site, but I have so > far never been able to ? they just time out? Is this just me or do o > thers have the same problem? > > > > Jan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/c632de80/attachment.html From stephen at mymessage.us Wed Jun 9 07:06:44 2010 From: stephen at mymessage.us (Stephen Cattaneo) Date: Wed, 9 Jun 2010 10:06:44 -0400 Subject: [Freeswitch-users] javascript bridge In-Reply-To: References: Message-ID: thank you for the responses. I will try these different methods. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print On Tue, Jun 8, 2010 at 1:32 PM, Samuel Macedo wrote: > Have you had any success with this bridge code? > Because I've had the same problem and my solution was: > > callee_dialstring = "{ignore_early_media=true,leg=2}sofia/user/XXX"; > session.execute('bridge', callee_dialstring); > > The problem here is that you have to establish the first leg before you can > dial the second number. > > Regards, > -- > Samuel Macedo > > On 7 June 2010 18:58, Michael Collins wrote: > >> I take it that this js is being called from the dialplan? If so then do >> what you need to do with outsession before you bridge it. Or consider using >> ESL and making a socket-based control program where you have much more >> flexibility... >> >> -MC >> >> On Sun, Jun 6, 2010 at 11:55 AM, stephen at stephenjc < >> stephen at stephenjc.com> wrote: >> >>> I have the following code, and after the bridge the javascript seems to >>> stop. >>> >>> outsession = new >>> Session("{ringback=\'%(2000,4000,440.0,480.0)\',instant_ringback=true,ignore_early_media=false}sofia/gateway/" >>> + providerhash["providername0"] + "/XXXXXXXXXXXX"); >>> bridge(session,outsession); >>> while(outsession.ready()) >>> { >>> console_log("notice","ping"); >>> } >>> >>> I am looking to manage the b leg, should i use execute_on_answer instead? >>> or is there a way to make the code continue after the bridge. >>> >>> Thanks, >>> Stephen C >>> -All of my email addresses go to the same place >>> -Save Paper, think before you print >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/35045bd0/attachment.html From jan.berger at video24.no Wed Jun 9 07:21:04 2010 From: jan.berger at video24.no (Jan Berger) Date: Wed, 9 Jun 2010 16:21:04 +0200 Subject: [Freeswitch-users] API pages on www.freeswitch.org In-Reply-To: <3ED7EAE4-C838-4634-885A-49DE83B346FB@freeswitch.org> References: <544C6ACCF81B47E3A7A3E41BD23F51E5@dell9400> <3ED7EAE4-C838-4634-885A-49DE83B346FB@freeswitch.org> Message-ID: <310172938E044E1FAE37FAD4B14BB1EC@dell9400> Hmmm - it works for me as well if I use Chrome rather than IE Should have guessed :-( _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins Sent: 9. juni 2010 16:03 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] API pages on www.freeswitch.org It's just a link to docs.freeswitch.org. I just checked it and worked ok for me. -MC Sent from my iPhone On Jun 9, 2010, at 6:19 AM, "Jan Berger" wrote: Hi, I try to access the API pages on the freeswitch site, but I have so far never been able to - they just time out? Is this just me or do others have the same problem? Jan _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/1ec70de4/attachment.html From fs-list at communicatefreely.net Wed Jun 9 07:43:57 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Wed, 09 Jun 2010 10:43:57 -0400 Subject: [Freeswitch-users] Freeswitch in an existing phone network In-Reply-To: <4C0E3FCE.1040803@iinet.net.au> References: <4C0E3FCE.1040803@iinet.net.au> Message-ID: <4C0FA8AD.8020907@communicatefreely.net> Hi Oliver, FreeSwitch will do whatever you tell it to do and no more. Here's a few suggestions though - Empty out the default dialplan directory. Don't throw those away, as you may want to reference them as examples, but move them somewhere else. Edit modules.conf.xml and comment or remove any modules that you don't need. This will also save memory and other resources. You can also disable all the SIP profiles except one. Pick one that makes sense (either Internal or External, it doesn't really matter that much), and edit it so that it makes sense with respect to your network. What is your topology? Will you just be setting freeswitch up with a static IP address and having calls sent to it by the main PBX? If that's the case, you can disable a lot of the STUN and uPNP functionality. Tell this profile to bind to the IP and port that the PBX will send the calls to. Then all you have to do, is create a very simple dialplan that will answer an incoming call and perform whatever task you want. You would essentially be starting with a blank sheet, adding just the functions that you want. Hope that makes sense. -Tim Oliver Schenk wrote: > Hi All, > > The company I work for currently has quite an extensive phone network > which gets carried between old analogue PABXes which also has an > interface to IP based phones. All the phones in our office are connected > via CAT5 cable using IP, however literally hundreds of phones out in the > field (we operate railway infrastructure) are on standard voice analogue > phones carried through fibreoptics. > > Anyway, I would like to use Freeswitch purely for its IVR and TTS > abilities and nothing else. So basically I just need it to act like a > slave to whatever IP phone network is already out there. I'm a bit > worried if I fire up freeswitch it will hijack the phone network! > > All our phones are accessible via a 5 digit extension. I would like > Freeswitch to be behind one of those ... say 12345. If anyone within our > phone network dials 12345 then Freeswitch should answer. I guess my > question is...how should I go about disabling most of FreeSwitch except > it's ability to pick up the phone and speak IVR/TTS and make an outgoing > call via the existing phone network? > > Any general pointers appreciated. > > > Thanks, > > Oliver Schenk > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 norm at voicenetwork.ca Tue Jun 8 09:22:00 2010 From: norm at voicenetwork.ca (Norman Tomlins) Date: Tue, 8 Jun 2010 12:22:00 -0400 Subject: [Freeswitch-users] Weekly FreeSwitch Conference call June 9 @ 1pm EST Message-ID: Weekly FreeSwitch Conference call June 9 @ 1pm EST Hi Everyone, I would just like to remind everyone that we have our weekly conference call on June , 9th @ 1pm EST. During this conference call Darren Schreiber will talk about mod_nibblebill. Information on mod_nilbble can be found here: http://wiki.freeswitch.org/wiki/Mod_nibblebill Information on Weekly Conference call: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_09 Norman Tomlins ---------------------------------------------------------------------------- Voice Network Inc. http://www.VoiceNetwork.ca Unlimited Incoming US DID's - $ 3.95 - No Setup Fees Unlimited Incoming Canadian DID's - $ 5.95 - No Setup Fees ---------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100608/94db2d39/attachment.html From bogdan.patrascoiu at sinergetic.ro Tue Jun 8 14:12:33 2010 From: bogdan.patrascoiu at sinergetic.ro (Bogdan Patrascoiu) Date: Wed, 9 Jun 2010 00:12:33 +0300 Subject: [Freeswitch-users] Post Hangup string Message-ID: Hello, I'm currently working on a FreeSWITCH box for a callcenter. One of the callcenter's requests was to rename each of the calls recordings based on the call's outcome, so each operator must tag each of it's call recording by hand. I planned to do this by ignoring both parties hangup requests and to catch a post end of conversation string from the voip client (in this case X-Lite) and fit this in the recording's file name by renaming it. I haven't done anything like it so far and wanted to ask if there is a standard approach for such post hangups implementations . Thanks, Bogdan Patrascoiu From ravriel_1 at yahoo.com Wed Jun 9 05:36:41 2010 From: ravriel_1 at yahoo.com (Ron Avriel) Date: Wed, 9 Jun 2010 05:36:41 -0700 (PDT) Subject: [Freeswitch-users] Anti-tromboning in FreeSWITCH? Message-ID: <313308.97850.qm@web45211.mail.sp1.yahoo.com> Hi, Is there any way to implement Anti-tromboning/Anti-Hairpinning/Media Release (http://en.wikipedia.org/wiki/Anti-tromboning) in FreeSWITCH? My scenario is similar to image in link above. Currently, when user A calls B I get two calls and media passing twice through FS. Is there any way for media to pass directly between A and B? Thanks, Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/8163e468/attachment.html From norm at voicenetwork.ca Wed Jun 9 08:03:52 2010 From: norm at voicenetwork.ca (Norman Tomlins) Date: Wed, 9 Jun 2010 11:03:52 -0400 Subject: [Freeswitch-users] Fwd: Weekly FreeSwitch Conference call June 9 @ 1pm EST In-Reply-To: References: Message-ID: Weekly FreeSwitch Conference call June 9 @ 1pm EST Hi Everyone, I would just like to remind everyone that we have our weekly conference call on June , 9th @ 1pm EST. During this conference call Darren Schreiber will talk about mod_nibblebill. Information on mod_nilbble can be found here: http://wiki.freeswitch.org/wiki/Mod_nibblebill Information on Weekly Conference call: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_09 Norman Tomlins ---------------------------------------------------------------------------- Voice Network Inc. http://www.VoiceNetwork.ca Unlimited Incoming US DID's - $ 3.95 - No Setup Fees Unlimited Incoming Canadian DID's - $ 5.95 - No Setup Fees ---------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/243de981/attachment.html From norm at voicenetwork.ca Wed Jun 9 08:12:23 2010 From: norm at voicenetwork.ca (Norman Tomlins) Date: Wed, 9 Jun 2010 11:12:23 -0400 Subject: [Freeswitch-users] Weekly FreeSwitch Conference call June 9 @ 1pm EST In-Reply-To: References: Message-ID: Weekly FreeSwitch Conference call Today June 9 @ 1pm EST Hi Everyone, I would just like to remind everyone that we have our weekly conference call on June , 9th @ 1pm EST. During this conference call Darren Schreiber will talk about mod_nibblebill. Information on mod_nilbble can be found here: http://wiki.freeswitch.org/wiki/Mod_nibblebill Information on Weekly Conference call: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_09 Norman Tomlins -------------------------------------------------------------------------------------------- Voice Network Inc. http://www.VoiceNetwork.ca Unlimited Incoming US DID's - $ 3.95 - No Setup Fees Unlimited Incoming Canadian DID's - $ 5.95 - No Setup Fees -------------------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/171ee9ae/attachment.html From testeador01 at gmail.com Wed Jun 9 08:17:26 2010 From: testeador01 at gmail.com (Milena) Date: Wed, 9 Jun 2010 10:17:26 -0500 Subject: [Freeswitch-users] Anti-tromboning in FreeSWITCH? In-Reply-To: <313308.97850.qm@web45211.mail.sp1.yahoo.com> References: <313308.97850.qm@web45211.mail.sp1.yahoo.com> Message-ID: http://wiki.freeswitch.org/wiki/Bypass_Media 2010/6/9 Ron Avriel > Hi, > > Is there any way to implement Anti-tromboning/Anti-Hairpinning/Media > Release (http://en.wikipedia.org/wiki/Anti-tromboning) in FreeSWITCH? > > My scenario is similar to image in link above. Currently, when user A calls > B I get two calls and media passing twice through FS. > Is there any way for media to pass directly between A and B? > > Thanks, > Ron > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/f0594aaf/attachment.html From david.ponzone at gmail.com Wed Jun 9 08:24:48 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Wed, 9 Jun 2010 17:24:48 +0200 Subject: [Freeswitch-users] Post Hangup string In-Reply-To: References: Message-ID: <027FB72A-AD72-4A11-B0CA-790C13A34893@gmail.com> Bogdan, There could be a way to do something nice if you don't hangup leg A after the call. The idea would be to disable hangup_after_bridge (it is by default I think), and to add a call to an internal IVR after the call to the customer is completed. In this IVR, you would listen to a wrap-up code entered as DTMF, and then you would rename to recording of the first call according to this. Some work to do, but should be doable. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 08/06/2010 ? 23:12, Bogdan Patrascoiu a ?crit : > Hello, > > I'm currently working on a FreeSWITCH box for a callcenter. One of the > callcenter's requests was to rename each of the calls recordings based > on the call's outcome, so each operator must tag each of it's call > recording by hand. I planned to do this by ignoring both parties > hangup requests and to catch a post end of conversation string from > the voip client (in this case X-Lite) and fit this in the recording's > file name by renaming it. > > I haven't done anything like it so far and wanted to ask if there is a > standard approach for such post hangups implementations . > > > Thanks, > > Bogdan Patrascoiu > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/5477ddc7/attachment-0001.html From robert.hadley at teotech.com Wed Jun 9 08:24:39 2010 From: robert.hadley at teotech.com (Robert Hadley) Date: Wed, 9 Jun 2010 08:24:39 -0700 Subject: [Freeswitch-users] application=record_session leaves a"session" around? In-Reply-To: References: Message-ID: Hi Brian, Yes, it was FreeSWITCH Version 1.0.head (git-d888ebc 2010-06-08 14-35-15 -0400) I will 'git' it today too. Thanks, Robert _____ From: Brian West [mailto:brian at freeswitch.org] Sent: Tuesday, June 08, 2010 4:58 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] application=record_session leaves a"session" around? Have you made sure you're on the very latest FreeSWITCH code? /b On Jun 8, 2010, at 6:44 PM, Robert Hadley wrote: Using seems to leave a "session" around. I notice this repeated message later when I restart sofia internal profile after recording calls: 2010-06-08 15:37:56.641151 [CRIT] sofia.c:1459 Waiting for 3 session(s) (Msg repeats 13 times) The number of sessions "Waiting for" is the number of calls I have recorded. Pastebin of console debug log at: http://pastebin.freeswitch.org/13157 Is this normal, a bug, or is there something I need to do after recording a call to free the session? Thanks, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/f0f6edd2/attachment.html From djbinter at gmail.com Wed Jun 9 08:58:40 2010 From: djbinter at gmail.com (DJB International) Date: Wed, 9 Jun 2010 08:58:40 -0700 Subject: [Freeswitch-users] Remote Party ID In-Reply-To: <005b01cb07da$aa53ba70$fefb2f50$@com> References: <014d01cb075a$37d55510$a77fff30$@com> <005b01cb07da$aa53ba70$fefb2f50$@com> Message-ID: It seems like you are doing it on gateway profile, not on sip profile like you mention in your first email. For gateway profile, you will need to change it to cid-type instead and you need to be at least on git-0152706. -djbinter On Wed, Jun 9, 2010 at 6:50 AM, Ken Fulmer < kenfulmer at icstechnologysolutions.com> wrote: > Thank you for replying and modifying the Wiki page. However, I?m not able > to get this to work. I tried the P-asserted identity as well with no luck. > > > > Here?s what I have configured for our connection to PaeTec: > > > > > > > > /// PAETEC /// > > /// account username *required* /// > > > > /// auth realm: *optional* same as gateway name, if blank /// > > > > /// username to use in from: *optional* same as username, if blank /// > > > > /// domain to use in from: *optional* same as realm, if blank /// > > > > /// account password *required* /// > > > > /// extension for inbound calls: *optional* same as username, if blank > /// > > > > /// proxy host: *optional* same as realm, if blank /// > > > > /// send register to this proxy: *optional* same as proxy, if blank /// > > > > /// expire in seconds: *optional* 3600, if blank /// > > > > /// register true / false/// > > > > /// which transport to use for register > > > > /// How many seconds before a retry when a failure or timeout occurs > > > > /// Use the callerid of an inbound call in the from field on outbound > calls via this gateway > > > > /// extra sip params to send in the contact > > > > /// send an options ping every x seconds, failure will unregister > and/or mark it down > > > > /// Replaces the "gw + external" in the Contact field with the > Extension value. > > > > /// Removes the default Remote Party ID. > > > > > > > > > > Thanks, > > > > Ken > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *DJB > International > *Sent:* Tuesday, June 08, 2010 6:24 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Remote Party ID > > > > > > > > > > > > > > > > > > > > > > > > > > On Tue, Jun 8, 2010 at 3:30 PM, Ken Fulmer < > kenfulmer at icstechnologysolutions.com> wrote: > > I?ve seen the following link but I?m still confused. > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type > > > > I realize this refers to channel variables, but is there a sip profile > parameter to remove Remote Party ID? > > > > Thanks, > > > > Ken > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100609/aa5af8d4/attachment-0001.html From anthony.minessale at gmail.com Wed Jun 9 09:13:22 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Jun 2010 11:13:22 -0500 Subject: [Freeswitch-users] Post Hangup string In-Reply-To: <027FB72A-AD72-4A11-B0CA-790C13A34893@gmail.com> References: <027FB72A-AD72-4A11-B0CA-790C13A34893@gmail.com> Message-ID: if you do that and the other end of the call hangs up first your application will not execute. FreeSWITCH does not continue in the dial-plan once a call has hungup. you are better off using the api_hangup_hook variable to have a script be called on hangup that processes the data. The channel is not available in the script but there is an event with all of the channel vars. On Wed, Jun 9, 2010 at 10:24 AM, David Ponzone wrote: > Bogdan, > > There could be a way to do something nice if you don't hangup leg A after > the call. > The idea would be to disable hangup_after_bridge (it is by default I > think), and to add a call to an internal IVR after the call to the customer > is completed. > In this IVR, you would listen to a wrap-up code entered as DTMF, and then > you would rename to recording of the first call according to this. > > Some work to do, but should be doable. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 08/06/2010 ? 23:12, Bogdan Patrascoiu a ?crit : > > Hello, > > I'm currently working on a FreeSWITCH box for a callcenter. One of the > callcenter's requests was to rename each of the calls recordings based > on the call's outcome, so each operator must tag each of it's call > recording by hand. I planned to do this by ignoring both parties > hangup requests and to catch a post end of conversation string from > the voip client (in this case X-Lite) and fit this in the recording's > file name by renaming it. > > I haven't done anything like it so far and wanted to ask if there is a > standard approach for such post hangups implementations . > > > Thanks, > > Bogdan Patrascoiu > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > -- 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/20100609/0f1eb15e/attachment.html From david.ponzone at gmail.com Wed Jun 9 09:32:45 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Wed, 9 Jun 2010 18:32:45 +0200 Subject: [Freeswitch-users] Post Hangup string In-Reply-To: References: <027FB72A-AD72-4A11-B0CA-790C13A34893@gmail.com> Message-ID: <5E052784-C615-4E37-BE57-02856E54043C@gmail.com> Tony, of course, you're right. In my mind, it was an outbound call center, so the agent will be leg A. But for an inbound call center, the agent would be leg B. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 09/06/2010 ? 18:13, Anthony Minessale a ?crit : > if you do that and the other end of the call hangs up first your > application will not execute. > FreeSWITCH does not continue in the dial-plan once a call has hungup. > > you are better off using the api_hangup_hook variable to have a > script be called on hangup that processes the data. > The channel is not available in the script but there is an event > with all of the channel vars. > > > On Wed, Jun 9, 2010 at 10:24 AM, David Ponzone > wrote: > Bogdan, > > There could be a way to do something nice if you don't hangup leg A > after the call. > The idea would be to disable hangup_after_bridge (it is by default I > think), and to add a call to an internal IVR after the call to the > customer is completed. > In this IVR, you would listen to a wrap-up code entered as DTMF, and > then you would rename to recording of the first call according to > this. > > Some work to do, but should be doable. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 08/06/2010 ? 23:12, Bogdan Patrascoiu a ?crit : > >> Hello, >> >> I'm currently working on a FreeSWITCH box for a callcenter. One of >> the >> callcenter's requests was to rename each of the calls recordings >> based >> on the call's outcome, so each operator must tag each of it's call >> recording by hand. I planned to do this by ignoring both parties >> hangup requests and to catch a post end of conversation string from >> the voip client (in this case X-Lite) and fit this in the recording's >> file name by renaming it. >> >> I haven't done anything like it so far and wanted to ask if there >> is a >> standard approach for such post hangups implementations . >> >> >> Thanks, >> >> Bogdan Patrascoiu >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > > > > -- > 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 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/91d827cc/attachment-0001.html From anthony.minessale at gmail.com Wed Jun 9 09:33:28 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Jun 2010 11:33:28 -0500 Subject: [Freeswitch-users] application=record_session leaves a"session" around? In-Reply-To: References: Message-ID: It's a bug but it's and edge case triggered by this: 2010-06-08 16:30:45.979138 [ERR] switch_ivr.c:1208 Can't re-establsh media on sofia/internal/1018 at 192.168.72.138:5060 This is because you appear to have bypass media enabled and you also are trying to record. When you do something that needs media on an unanswered channel it will try to establish media. But since the call is not answered the attempt will fail. You should not mix bypass media and recording in this way but I committed a fix for the stuck channel. On Wed, Jun 9, 2010 at 10:24 AM, Robert Hadley wrote: > Hi Brian, > > > > Yes, it was FreeSWITCH Version 1.0.head (git-d888ebc 2010-06-08 14-35-15 > -0400) > > > > I will ?git? it today too. > > > > Thanks, > > Robert > > > ------------------------------ > > *From:* Brian West [mailto:brian at freeswitch.org] > *Sent:* Tuesday, June 08, 2010 4:58 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] application=record_session leaves > a"session" around? > > > > Have you made sure you're on the very latest FreeSWITCH code? > > > > /b > > > > On Jun 8, 2010, at 6:44 PM, Robert Hadley wrote: > > > > Using seems > to leave a ?session? around. > > I notice this repeated message later when I restart sofia internal profile > after recording calls: > 2010-06-08 15:37:56.641151 [CRIT] sofia.c:1459 Waiting for 3 session(s) > (Msg repeats 13 times) > > The number of sessions ?Waiting for? is the number of calls I have > recorded. > > Pastebin of console debug log at: http://pastebin.freeswitch.org/13157 > > > > Is this normal, a bug, or is there something I need to do after recording > a call to free the session? > > > > Thanks, > > Robert > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100609/685ea6c2/attachment.html From ravriel_1 at yahoo.com Wed Jun 9 11:06:55 2010 From: ravriel_1 at yahoo.com (Ron Avriel) Date: Wed, 9 Jun 2010 11:06:55 -0700 (PDT) Subject: [Freeswitch-users] Anti-tromboning in FreeSWITCH? In-Reply-To: References: <313308.97850.qm@web45211.mail.sp1.yahoo.com> Message-ID: <101954.30288.qm@web45203.mail.sp1.yahoo.com> Bypass media only connects media of a single bridged call. The problem here is how to connect media between two different calls. Ron ________________________________ From: Milena To: freeswitch-users at lists.freeswitch.org Sent: Wed, June 9, 2010 6:17:26 PM Subject: Re: [Freeswitch-users] Anti-tromboning in FreeSWITCH? http://wiki.freeswitch.org/wiki/Bypass_Media 2010/6/9 Ron Avriel Hi, > >Is there any way to implement Anti-tromboning/Anti-Hairpinning/Media Release (http://en.wikipedia.org/wiki/Anti-tromboning) in FreeSWITCH? > >My scenario is similar to image in link above. Currently, when user A calls B I get two calls and media passing twice through FS. >Is there any way for media to pass directly between A and B? > >Thanks, >Ron > > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/a946d6cc/attachment.html From codeghar at gmail.com Wed Jun 9 11:33:12 2010 From: codeghar at gmail.com (Code Ghar) Date: Wed, 9 Jun 2010 13:33:12 -0500 Subject: [Freeswitch-users] Weekly FreeSwitch Conference call June 9 @ 1pm EST In-Reply-To: References: Message-ID: In today's conference call an idea was discussed to have a presentation on Wireshark. I could never call myself an expert but I have done a lot of troubleshooting using Wireshark with multiple carriers. I have even trained a couple of my co-workers in using Wireshark to troubleshoot SIP issues. I would like to volunteer for this presentation. Since I haven't done anything like this before, I would need help in preparing for it. And any specific questions anyone may have which they would like discussed would be great too. On Tue, Jun 8, 2010 at 11:22 AM, Norman Tomlins wrote: > Weekly FreeSwitch Conference call June 9 @ 1pm EST > > > > Hi Everyone, > > > > I would just like to remind everyone that we have our weekly conference > call on June , 9th @ 1pm EST. > > > During this conference call Darren Schreiber will talk about > mod_nibblebill. > > > Information on mod_nilbble can be found here: > http://wiki.freeswitch.org/wiki/Mod_nibblebill > > Information on Weekly Conference call: > http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_09 > > > > Norman Tomlins > > > > ---------------------------------------------------------------------------- > > Voice Network Inc. > > http://www.VoiceNetwork.ca > > Unlimited Incoming US DID's - $ 3.95 - No Setup Fees > > Unlimited Incoming Canadian DID's - $ 5.95 - No Setup Fees > > > ---------------------------------------------------------------------------- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/8f8b1ec6/attachment.html From anthony.minessale at gmail.com Wed Jun 9 11:34:33 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Jun 2010 13:34:33 -0500 Subject: [Freeswitch-users] Anti-tromboning in FreeSWITCH? In-Reply-To: <101954.30288.qm@web45203.mail.sp1.yahoo.com> References: <313308.97850.qm@web45211.mail.sp1.yahoo.com> <101954.30288.qm@web45203.mail.sp1.yahoo.com> Message-ID: try having sip_auto_simplify channel var set to true at the time of the bridge. On Wed, Jun 9, 2010 at 1:06 PM, Ron Avriel wrote: > Bypass media only connects media of a single bridged call. > The problem here is how to connect media between two different calls. > > Ron > > ------------------------------ > *From:* Milena > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Wed, June 9, 2010 6:17:26 PM > *Subject:* Re: [Freeswitch-users] Anti-tromboning in FreeSWITCH? > > http://wiki.freeswitch.org/wiki/Bypass_Media > > 2010/6/9 Ron Avriel > >> Hi, >> >> Is there any way to implement Anti-tromboning/Anti-Hairpinning/Media >> Release (http://en.wikipedia.org/wiki/Anti-tromboning) in FreeSWITCH? >> >> My scenario is similar to image in link above. Currently, when user A >> calls B I get two calls and media passing twice through FS. >> Is there any way for media to pass directly between A and B? >> >> Thanks, >> Ron >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100609/8d8d7d7e/attachment-0001.html From dome at tel.co.th Wed Jun 9 11:41:14 2010 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 10 Jun 2010 01:41:14 +0700 Subject: [Freeswitch-users] api_hangup_hook and hupall Message-ID: I use api_hangup_hook for do something when call hangup. but when i use hupall command from CLI api_hangup_hook do nothing. Is it bug ? Dome C. From Prometheus001 at gmx.net Wed Jun 9 11:44:09 2010 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 09 Jun 2010 20:44:09 +0200 Subject: [Freeswitch-users] Sending NOTIFY message to a phone via XML-RPC In-Reply-To: References: <4C08C995.8060500@gmx.net> <4C0D24B0.3010708@gmx.net> Message-ID: <4C0FE0F9.4010102@gmx.net> Thanks Anthony with event_socket I got it working. Best regards Peter Anthony Minessale schrieb: > you can't do sendevent over XML > its a command for event_socket > > it would probably be possible to make such a thing but it does not > currently exist. > > > On Mon, Jun 7, 2010 at 11:56 AM, Peter P GMX > wrote: > > Nobody has a working XML sample for sending NOTIFY events? > The wiki is quite lean on this. I will update the wiki with a working > sample as soon as I have one. > > So any suggestions? > > Best regards > Peter > > Peter P GMX schrieb: > > Hello, > > > > I try to send a NOTIFY message from my application to a > registered phone > > via XML-RPC. > > This is the XML which is sent: > > > > > > > > freeswitch.api > > > > > > > > sendevent > > > > > > > > > > > NOTIFY,profile=internal,event-string=check-sync;reboot=false,user=200,host=192.168.178.220,content-type=application/simple-message-summary > > > > > > > > > > > > However I receive an error message: > > . > > . > > . > > ERROR!. > > . > > . > > > > I also tried to send the data as a struct but FS complains about > needing > > a string as parameters. > > > > What am I doing wrong? Anybody has a valid sample XML for a > NOTIFY message? > > > > Best regards > > Peter > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > > > > > -- > 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 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sos at sokhapkin.dyndns.org Wed Jun 9 11:57:26 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 9 Jun 2010 14:57:26 -0400 Subject: [Freeswitch-users] Anti-tromboning in FreeSWITCH? In-Reply-To: References: <313308.97850.qm@web45211.mail.sp1.yahoo.com> <101954.30288.qm@web45203.mail.sp1.yahoo.com> Message-ID: <201006091457.26143.sos@sokhapkin.dyndns.org> What sip_auto_simplify settings does? On Wednesday 09 June 2010, Anthony Minessale wrote: > try having sip_auto_simplify channel var set to true at the time of the > bridge. > > On Wed, Jun 9, 2010 at 1:06 PM, Ron Avriel wrote: > > Bypass media only connects media of a single bridged call. > > The problem here is how to connect media between two different calls. > > > > Ron > > > > ------------------------------ > > *From:* Milena > > *To:* freeswitch-users at lists.freeswitch.org > > *Sent:* Wed, June 9, 2010 6:17:26 PM > > *Subject:* Re: [Freeswitch-users] Anti-tromboning in FreeSWITCH? > > > > http://wiki.freeswitch.org/wiki/Bypass_Media > > > > 2010/6/9 Ron Avriel > > > >> Hi, > >> > >> Is there any way to implement Anti-tromboning/Anti-Hairpinning/Media > >> Release (http://en.wikipedia.org/wiki/Anti-tromboning) in FreeSWITCH? > >> > >> My scenario is similar to image in link above. Currently, when user A > >> calls B I get two calls and media passing twice through FS. > >> Is there any way for media to pass directly between A and B? > >> > >> Thanks, > >> Ron > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.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 > From anthony.minessale at gmail.com Wed Jun 9 12:07:26 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Jun 2010 14:07:26 -0500 Subject: [Freeswitch-users] Anti-tromboning in FreeSWITCH? In-Reply-To: <201006091457.26143.sos@sokhapkin.dyndns.org> References: <313308.97850.qm@web45211.mail.sp1.yahoo.com> <101954.30288.qm@web45203.mail.sp1.yahoo.com> <201006091457.26143.sos@sokhapkin.dyndns.org> Message-ID: if it detects a bridge in this trombone scenario it will use REFER to transfer the 2 legs on the far end box to be bridged like an attended transfer. On Wed, Jun 9, 2010 at 1:57 PM, Sergey Okhapkin wrote: > What sip_auto_simplify settings does? > > On Wednesday 09 June 2010, Anthony Minessale wrote: > > try having sip_auto_simplify channel var set to true at the time of the > > bridge. > > > > On Wed, Jun 9, 2010 at 1:06 PM, Ron Avriel wrote: > > > Bypass media only connects media of a single bridged call. > > > The problem here is how to connect media between two different calls. > > > > > > Ron > > > > > > ------------------------------ > > > *From:* Milena > > > *To:* freeswitch-users at lists.freeswitch.org > > > *Sent:* Wed, June 9, 2010 6:17:26 PM > > > *Subject:* Re: [Freeswitch-users] Anti-tromboning in FreeSWITCH? > > > > > > http://wiki.freeswitch.org/wiki/Bypass_Media > > > > > > 2010/6/9 Ron Avriel > > > > > >> Hi, > > >> > > >> Is there any way to implement Anti-tromboning/Anti-Hairpinning/Media > > >> Release (http://en.wikipedia.org/wiki/Anti-tromboning) in FreeSWITCH? > > >> > > >> My scenario is similar to image in link above. Currently, when user A > > >> calls B I get two calls and media passing twice through FS. > > >> Is there any way for media to pass directly between A and B? > > >> > > >> Thanks, > > >> Ron > > >> > > >> > > >> _______________________________________________ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.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 > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100609/d684cfc4/attachment.html From dome at tel.co.th Wed Jun 9 12:30:08 2010 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 10 Jun 2010 02:30:08 +0700 Subject: [Freeswitch-users] api_hangup_hook and hupall In-Reply-To: References: Message-ID: Sorry it's work fine. bug in my script :) 2010/6/10 Dome Charoenyost : > I ?use api_hangup_hook for do something when call hangup. but when i > use hupall command from CLI api_hangup_hook do nothing. > Is it bug ? > > Dome C. > From mgende at gendesign.com Wed Jun 9 12:56:02 2010 From: mgende at gendesign.com (Michael Gende) Date: Wed, 9 Jun 2010 14:56:02 -0500 Subject: [Freeswitch-users] Out-Going Call Transfer Fixed Message-ID: Hello, Just to follow up on my own follow-up, we've fixed our "we can't transfer an outgoing call to another local extension" problem. Turned out, the problem was in the dial plan, the local_extension extension, as one might expect. We added a line to set the local domain properly and all was well. Now, we see outbound called transfers going to, for example, 1001 at the.real.lan.ip.of.the.device instead of 1001 at the.wan.ip.address.Naturally, the former works and the latter don't. Hooray. Also, many thanks to the intrepid ones who asked me for more info and tried to offer some advice. Regards, Mike G. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/e24d9b2c/attachment.html From robert.hadley at teotech.com Wed Jun 9 13:04:36 2010 From: robert.hadley at teotech.com (Robert Hadley) Date: Wed, 9 Jun 2010 13:04:36 -0700 Subject: [Freeswitch-users] application=record_session leaves a"session"around? In-Reply-To: References: Message-ID: <90A5795028D44067AA1B28C9A68E4469@greyhawk.tonecommander.com> Thanks Anthony, Yes, I have inbound-bypass-media enabled globally (and late-negotiation). I added to dialplan when recording but I still get the same error. 2010-06-09 12:50:04.877331 [ERR] switch_ivr.c:1208 Can't re-establsh media on sofia/internal/1018 at 192.168.72.138:5060 Is it okay to disable bypass_media on a per-call basis this way when recording, to override the global setting? Thanks, Robert _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Wednesday, June 09, 2010 9:33 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] application=record_session leaves a"session"around? It's a bug but it's and edge case triggered by this: 2010-06-08 16:30:45.979138 [ERR] switch_ivr.c:1208 Can't re-establsh media on sofia/internal/1018 at 192.168.72.138:5060 This is because you appear to have bypass media enabled and you also are trying to record. When you do something that needs media on an unanswered channel it will try to establish media. But since the call is not answered the attempt will fail. You should not mix bypass media and recording in this way but I committed a fix for the stuck channel. On Wed, Jun 9, 2010 at 10:24 AM, Robert Hadley wrote: Hi Brian, Yes, it was FreeSWITCH Version 1.0.head (git-d888ebc 2010-06-08 14-35-15 -0400) I will 'git' it today too. Thanks, Robert _____ From: Brian West [mailto:brian at freeswitch.org] Sent: Tuesday, June 08, 2010 4:58 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] application=record_session leaves a"session" around? Have you made sure you're on the very latest FreeSWITCH code? /b On Jun 8, 2010, at 6:44 PM, Robert Hadley wrote: Using seems to leave a "session" around. I notice this repeated message later when I restart sofia internal profile after recording calls: 2010-06-08 15:37:56.641151 [CRIT] sofia.c:1459 Waiting for 3 session(s) (Msg repeats 13 times) The number of sessions "Waiting for" is the number of calls I have recorded. Pastebin of console debug log at: http://pastebin.freeswitch.org/13157 Is this normal, a bug, or is there something I need to do after recording a call to free the session? Thanks, Robert _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20100609/cca398a8/attachment-0001.html From anthony.minessale at gmail.com Wed Jun 9 13:13:46 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Jun 2010 15:13:46 -0500 Subject: [Freeswitch-users] application=record_session leaves a"session"around? In-Reply-To: <90A5795028D44067AA1B28C9A68E4469@greyhawk.tonecommander.com> References: <90A5795028D44067AA1B28C9A68E4469@greyhawk.tonecommander.com> Message-ID: the variable can't reverse the profile param if you want to pick and choose you need to turn it off in the profile and only use the var. On Wed, Jun 9, 2010 at 3:04 PM, Robert Hadley wrote: > Thanks Anthony, > > > > Yes, I have inbound-bypass-media enabled globally (and late-negotiation). > > > > I added to dialplan > when recording but I still get the same error. > > 2010-06-09 12:50:04.877331 [ERR] switch_ivr.c:1208 Can't re-establsh media > on sofia/internal/1018 at 192.168.72.138:5060 > > > > Is it okay to disable bypass_media on a per-call basis this way when > recording, to override the global setting? > > > > Thanks, > > Robert > > > ------------------------------ > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Wednesday, June 09, 2010 9:33 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] application=record_session leaves > a"session"around? > > > > It's a bug but it's and edge case triggered by this: > > > > 2010-06-08 16:30:45.979138 [ERR] switch_ivr.c:1208 Can't re-establsh media > on sofia/internal/1018 at 192.168.72.138:5060 > > > > This is because you appear to have bypass media enabled and you also are > trying to record. > > When you do something that needs media on an unanswered channel it will try > to establish media. > > But since the call is not answered the attempt will fail. > > > > You should not mix bypass media and recording in this way but I committed a > fix for the stuck channel. > > > > > > On Wed, Jun 9, 2010 at 10:24 AM, Robert Hadley > wrote: > > Hi Brian, > > > > Yes, it was FreeSWITCH Version 1.0.head (git-d888ebc 2010-06-08 14-35-15 > -0400) > > > > I will ?git? it today too. > > > > Thanks, > > Robert > > > ------------------------------ > > *From:* Brian West [mailto:brian at freeswitch.org] > *Sent:* Tuesday, June 08, 2010 4:58 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] application=record_session leaves > a"session" around? > > > > Have you made sure you're on the very latest FreeSWITCH code? > > > > /b > > > > On Jun 8, 2010, at 6:44 PM, Robert Hadley wrote: > > > > Using seems to > leave a ?session? around. > > I notice this repeated message later when I restart sofia internal profile > after recording calls: > 2010-06-08 15:37:56.641151 [CRIT] sofia.c:1459 Waiting for 3 session(s) > (Msg repeats 13 times) > > The number of sessions ?Waiting for? is the number of calls I have > recorded. > > Pastebin of console debug log at: http://pastebin.freeswitch.org/13157 > > > > Is this normal, a bug, or is there something I need to do after recording > a call to free the session? > > > > Thanks, > > Robert > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100609/de378d35/attachment.html From telteclistas at gmail.com Wed Jun 9 13:47:05 2010 From: telteclistas at gmail.com (leonardo alves) Date: Wed, 9 Jun 2010 16:47:05 -0400 Subject: [Freeswitch-users] Early media configuration In-Reply-To: References: Message-ID: The ignore_early_media value you use in the bridge command, like this. As for the no audio is probably due to a nat in your network. Leonardo On Tue, Jun 8, 2010 at 5:11 PM, Sean Holt wrote: > Hello lists, > > I?m having 2 issues and wondering if someone can help. > > Here are my specs > Most recent git from freeswitch > Centos 5 > > I?ve include my dialplan > http://pastebin.freeswitch.org/13156 > > 1st: > When calling any endpoint that requires digit input, the endpoint does not > recognize the dtmf. I?ve done a wireshark trace and I see the dtmf output, > so I know that part is working. Tech support at Bandwidth.com says I need > to enable early media, so I added the ignore_early_media=true in my dialplan > but still no success. I?m thinking I don?t have ignore early media in the > right place, but I?ve included my dialplan for viewing. > > 2nd: > Incoming calls have no audio after the call bridges. Hard to say what the > issue is, but it might have something to do with early media. > > Thoughts? > > Thanks > Sean > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/650d1eef/attachment-0001.html From macedoslm at gmail.com Wed Jun 9 14:19:28 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Wed, 9 Jun 2010 18:19:28 -0300 Subject: [Freeswitch-users] Gateway Registration In-Reply-To: References: Message-ID: How can I set the gateway domain? Can it be different from the gateway name? Example: *Gateway Name*: gateway.com *Gateway Domain*: gateway.com Thanks, -- Samuel Macedo On 8 June 2010 13:03, Samuel Macedo wrote: > But the name of the gateway is the domain that will be used to register, or > no? > > -- > Samuel Macedo > > > On 8 June 2010 06:46, Steven Ayre wrote: > >> Configure the same gateway twice, with different names. Which gateway name >> you use will determine which username is used. >> >> >> >> On 7 June 2010 20:26, Samuel Macedo wrote: >> >>> Hi, >>> >>> I want to use the same Gateway but with different users and passwords to >>> make and receive calls. >>> This is the scenario: >>> I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch >>> UserAgents (Sofia External) to this gateway so I can receive inbound calls. >>> And to make a Outbound call I want to decide witch UserAgent I will use. >>> >>> Is there anyway to do this? >>> >>> Regards, >>> -- >>> Samuel Macedo >>> Belo Horizonte - Brazil >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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/20100609/aed3cbed/attachment.html From gcd at i.ph Wed Jun 9 15:48:44 2010 From: gcd at i.ph (Nandy Dagondon) Date: Thu, 10 Jun 2010 06:48:44 +0800 Subject: [Freeswitch-users] Gateway Registration In-Reply-To: References: Message-ID: you can set different names for the gateway name but the domain must be the same "gateway.com" . . . . . . -nandy On Thu, Jun 10, 2010 at 5:19 AM, Samuel Macedo wrote: > How can I set the gateway domain? Can it be different from the gateway > name? > Example: > > > > > > > > > > *Gateway Name*: gateway.com > *Gateway Domain*: gateway.com > > Thanks, > -- > Samuel Macedo > > > On 8 June 2010 13:03, Samuel Macedo wrote: > >> But the name of the gateway is the domain that will be used to register, >> or no? >> >> -- >> Samuel Macedo >> >> >> On 8 June 2010 06:46, Steven Ayre wrote: >> >>> Configure the same gateway twice, with different names. Which gateway >>> name you use will determine which username is used. >>> >>> >>> >>> On 7 June 2010 20:26, Samuel Macedo wrote: >>> >>>> Hi, >>>> >>>> I want to use the same Gateway but with different users and passwords to >>>> make and receive calls. >>>> This is the scenario: >>>> I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch >>>> UserAgents (Sofia External) to this gateway so I can receive inbound calls. >>>> And to make a Outbound call I want to decide witch UserAgent I will use. >>>> >>>> Is there anyway to do this? >>>> >>>> Regards, >>>> -- >>>> Samuel Macedo >>>> Belo Horizonte - Brazil >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/4a7aacc1/attachment.html From dswardstrom at remotelink.com Wed Jun 9 16:05:03 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Wed, 9 Jun 2010 18:05:03 -0500 (CDT) Subject: [Freeswitch-users] JavaScript control Transfer In-Reply-To: <2144151915.6.1276124380614.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <760778166.8.1276124703365.JavaMail.root@srvr12.remotelinkml.com> I have been working on a small Javascript application to determine the specific Javascript application to execute based on the DNIS. The goal is: * Provide an entry in a dialplan public XML file to invoke the application. * Use a database lookup (ODBC) using DNIS. * Fail if the DNIS is not found (hangup the session) * If the DNIS is found: ** use another entry in the database to determine ** the working application and any static parameters. * Invoke/Transfer to/Activate the other application. It is fairly simple to do all but the first step. I have been having problems trying to get the last step to work. What I would like is something like this: session.execute("transfer", transfer_string); Where transfer_string is the string from the Database. Or where transfer_string is the string from the DB plus some fixed options. I would prefer not to have to add entries into the dialplan/pubic/00_xx.xml file but will if I have to. I did notice an option called "inline" which potentially avoid this. I suspect part of my problem is not quite understanding the transfer syntax. If I ever get this running properly, I will provide a Wiki example of a simplified version of this (ie, without some of work specific code). Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom From msc at freeswitch.org Wed Jun 9 16:14:12 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Jun 2010 16:14:12 -0700 Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1276001266045-5153498.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> <1275859376110-5146705.post@n2.nabble.com> <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> <1275862175540-5146803.post@n2.nabble.com> <1276001266045-5153498.post@n2.nabble.com> Message-ID: On Tue, Jun 8, 2010 at 5:47 AM, peely wrote: > > Hi again, > > I've checked with BT, and they are unable to support reinvites with a > Require: timer. > > Reading the RFC, it seems that new transactions hsould not have the > Require: > timer but should revert to Supported: timer. > > Which RFC is that? And which section? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/fddf906a/attachment.html From anthony.minessale at gmail.com Wed Jun 9 16:20:29 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Jun 2010 18:20:29 -0500 Subject: [Freeswitch-users] JavaScript control Transfer In-Reply-To: <760778166.8.1276124703365.JavaMail.root@srvr12.remotelinkml.com> References: <2144151915.6.1276124380614.JavaMail.root@srvr12.remotelinkml.com> <760778166.8.1276124703365.JavaMail.root@srvr12.remotelinkml.com> Message-ID: when you say execute an application do you mean literally a dp application like set or voicemail ? transfer takes an extension as an arg not an application and when you do run that transfer command you must exit your script so the transfer can take place. session.execute("transfer", ""); // control returns here or if you want to run an extension like an app and return to your script do session.execute("execute_extension", ""); // control returns here On Wed, Jun 9, 2010 at 6:05 PM, David Swardstrom wrote: > I have been working on a small Javascript application to determine > the specific Javascript application to execute based on the DNIS. > The goal is: > * Provide an entry in a dialplan public XML file to invoke the application. > * Use a database lookup (ODBC) using DNIS. > * Fail if the DNIS is not found (hangup the session) > * If the DNIS is found: > ** use another entry in the database to determine > ** the working application and any static parameters. > * Invoke/Transfer to/Activate the other application. > > It is fairly simple to do all but the first step. > I have been having problems trying to get the last step to work. > > What I would like is something like this: > session.execute("transfer", transfer_string); > Where transfer_string is the string from the Database. > Or where transfer_string is the string from the DB > plus some fixed options. > > I would prefer not to have to add entries into > the dialplan/pubic/00_xx.xml file but will if I have to. > I did notice an option called "inline" which potentially avoid this. > > I suspect part of my problem is not quite understanding the transfer > syntax. > > If I ever get this running properly, I will provide a Wiki example of a > simplified version of this (ie, without some of work specific code). > > Regards, > David Swardstrom > (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100609/372cba27/attachment-0001.html From msc at freeswitch.org Wed Jun 9 16:20:48 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Jun 2010 16:20:48 -0700 Subject: [Freeswitch-users] Call Forwarding/Diversion Dialplan In-Reply-To: References: Message-ID: Check out this discussion thread and see if it answers your question... http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/057261.html -MC On Tue, Jun 8, 2010 at 1:08 PM, Michael De Lorenzo < delorenzodesign at gmail.com> wrote: > I'm trying to write a dialplan that will forward a call to a PSTN number. > > PSTN (Party A) Inbound call to your Freeswitch (Party B), which is > forwarded out to the PSTN (Party C) > > I've found examples for transferring calls to other Freeswitch users, but I > haven't been able to get this to work properly--where it sets a Diversion > header. > > Can anyone help? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/20982182/attachment.html From anthony.minessale at gmail.com Wed Jun 9 16:22:05 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Jun 2010 18:22:05 -0500 Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> <1275859376110-5146705.post@n2.nabble.com> <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> <1275862175540-5146803.post@n2.nabble.com> <1276001266045-5153498.post@n2.nabble.com> Message-ID: and when did the subject change? is this a hijacked thread? On Wed, Jun 9, 2010 at 6:14 PM, Michael Collins wrote: > > > On Tue, Jun 8, 2010 at 5:47 AM, peely wrote: > >> >> Hi again, >> >> I've checked with BT, and they are unable to support reinvites with a >> Require: timer. >> >> Reading the RFC, it seems that new transactions hsould not have the >> Require: >> timer but should revert to Supported: timer. >> >> Which RFC is that? And which section? > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100609/fe7cbce0/attachment.html From msc at freeswitch.org Wed Jun 9 16:26:22 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Jun 2010 16:26:22 -0700 Subject: [Freeswitch-users] Gateway Registration In-Reply-To: References: Message-ID: On Wed, Jun 9, 2010 at 2:19 PM, Samuel Macedo wrote: > How can I set the gateway domain? Can it be different from the gateway > name? Yes, you can use the "realm" param. See the example gateway config in conf/sip_profiles/external/example.xml -MC > Example: > > > > > > > > > > *Gateway Name*: gateway.com > *Gateway Domain*: gateway.com > > Thanks, > -- > Samuel Macedo > > > On 8 June 2010 13:03, Samuel Macedo wrote: > >> But the name of the gateway is the domain that will be used to register, >> or no? >> >> -- >> Samuel Macedo >> >> >> On 8 June 2010 06:46, Steven Ayre wrote: >> >>> Configure the same gateway twice, with different names. Which gateway >>> name you use will determine which username is used. >>> >>> >>> >>> On 7 June 2010 20:26, Samuel Macedo wrote: >>> >>>> Hi, >>>> >>>> I want to use the same Gateway but with different users and passwords to >>>> make and receive calls. >>>> This is the scenario: >>>> I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch >>>> UserAgents (Sofia External) to this gateway so I can receive inbound calls. >>>> And to make a Outbound call I want to decide witch UserAgent I will use. >>>> >>>> Is there anyway to do this? >>>> >>>> Regards, >>>> -- >>>> Samuel Macedo >>>> Belo Horizonte - Brazil >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/06617f22/attachment.html From telteclistas at gmail.com Wed Jun 9 16:34:30 2010 From: telteclistas at gmail.com (leonardo alves) Date: Wed, 9 Jun 2010 19:34:30 -0400 Subject: [Freeswitch-users] Gateway Registration In-Reply-To: References: Message-ID: No you register 2 gateways. And in the dial plan you can do the bridge with the gateway you want using the name of the gateway. Leonardo On Wed, Jun 9, 2010 at 5:19 PM, Samuel Macedo wrote: > How can I set the gateway domain? Can it be different from the gateway > name? > Example: > > > > > > > > > > *Gateway Name*: gateway.com > *Gateway Domain*: gateway.com > > Thanks, > -- > Samuel Macedo > > > On 8 June 2010 13:03, Samuel Macedo wrote: > >> But the name of the gateway is the domain that will be used to register, >> or no? >> >> -- >> Samuel Macedo >> >> >> On 8 June 2010 06:46, Steven Ayre wrote: >> >>> Configure the same gateway twice, with different names. Which gateway >>> name you use will determine which username is used. >>> >>> >>> >>> On 7 June 2010 20:26, Samuel Macedo wrote: >>> >>>> Hi, >>>> >>>> I want to use the same Gateway but with different users and passwords to >>>> make and receive calls. >>>> This is the scenario: >>>> I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch >>>> UserAgents (Sofia External) to this gateway so I can receive inbound calls. >>>> And to make a Outbound call I want to decide witch UserAgent I will use. >>>> >>>> Is there anyway to do this? >>>> >>>> Regards, >>>> -- >>>> Samuel Macedo >>>> Belo Horizonte - Brazil >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100609/76b8520d/attachment-0001.html From msc at freeswitch.org Wed Jun 9 16:40:17 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Jun 2010 16:40:17 -0700 Subject: [Freeswitch-users] Freeswitch in an existing phone network In-Reply-To: <4C0FA8AD.8020907@communicatefreely.net> References: <4C0E3FCE.1040803@iinet.net.au> <4C0FA8AD.8020907@communicatefreely.net> Message-ID: Tim and Joseph make good points. You start with the default configuration and then remove what you do not need. Frankly, the default configuration is just fine for you. FS won't "hijack" your VoIP infrastructure so don't sweat that. It's simply a server sitting on your network waiting for calls. My recommendation would be to do a basic install (see the wiki) and then make one addition to the public.xml dialplan file. Add this simple extension before the closing tag: Press F6 or type "reloadxml" at the FreeSWITCH command line. Now if you throw anonymous calls at this server it will handle four-digit extensions. For example, if you call sip:9999 at x.x.x.x:5060 then you should hear music on hold. Call sip:5000 at x.x.x.x:5060 and you should hear the demo IVR. (x.x.x.x = FS server IP address) This will let you test a lot of FS features without having to register phones to your FS server, although you are certainly free to do that. Be sure to read the newbie article that I wrote for Linux-Pro magazine last year: http://bit.ly/EpVrv It will help you get up and running quickly. (Note: we switched to git from svn, so we recommend you use git to download the source...) Have fun! -MC On Wed, Jun 9, 2010 at 7:43 AM, Tim St. Pierre < fs-list at communicatefreely.net> wrote: > Hi Oliver, > > FreeSwitch will do whatever you tell it to do and no more. > > Here's a few suggestions though - > > Empty out the default dialplan directory. Don't throw those away, as you > may want to reference them > as examples, but move them somewhere else. > > Edit modules.conf.xml and comment or remove any modules that you don't > need. This will also save > memory and other resources. > > You can also disable all the SIP profiles except one. Pick one that makes > sense (either Internal or > External, it doesn't really matter that much), and edit it so that it makes > sense with respect to > your network. What is your topology? Will you just be setting freeswitch > up with a static IP > address and having calls sent to it by the main PBX? If that's the case, > you can disable a lot of > the STUN and uPNP functionality. Tell this profile to bind to the IP and > port that the PBX will > send the calls to. > > Then all you have to do, is create a very simple dialplan that will answer > an incoming call and > perform whatever task you want. You would essentially be starting with a > blank sheet, adding just > the functions that you want. > > Hope that makes sense. > > -Tim > > Oliver Schenk wrote: > > Hi All, > > > > The company I work for currently has quite an extensive phone network > > which gets carried between old analogue PABXes which also has an > > interface to IP based phones. All the phones in our office are connected > > via CAT5 cable using IP, however literally hundreds of phones out in the > > field (we operate railway infrastructure) are on standard voice analogue > > phones carried through fibreoptics. > > > > Anyway, I would like to use Freeswitch purely for its IVR and TTS > > abilities and nothing else. So basically I just need it to act like a > > slave to whatever IP phone network is already out there. I'm a bit > > worried if I fire up freeswitch it will hijack the phone network! > > > > All our phones are accessible via a 5 digit extension. I would like > > Freeswitch to be behind one of those ... say 12345. If anyone within our > > phone network dials 12345 then Freeswitch should answer. I guess my > > question is...how should I go about disabling most of FreeSwitch except > > it's ability to pick up the phone and speak IVR/TTS and make an outgoing > > call via the existing phone network? > > > > Any general pointers appreciated. > > > > > > Thanks, > > > > Oliver Schenk > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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/20100609/a4ceb6b3/attachment.html From dswardstrom at remotelink.com Wed Jun 9 17:14:25 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Wed, 9 Jun 2010 17:14:25 -0700 (PDT) Subject: [Freeswitch-users] JavaScript control Transfer In-Reply-To: References: <760778166.8.1276124703365.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <1276128865845-5161043.post@n2.nabble.com> I really want to do the transfer and plan to exit() after the transfer takes place. What I have is a string like the following: "conf-ivr.js 6". The conf-ivr.js is my version of the conf-ivr.js example. The value "6" indicates a sub-type of conference so that we can support different flavors. I guess I could use the following: session.execute("transfer", "conf-ivr6"); Then put the following into the xml: What I hoped for was a way to avoid putting specific stuff for each application into the XML. I just wanted to be able to provide a new JS application or a new flavor of a JS application without having to also modify the XML and get the XML reloaded into FreeSwitch. Is it possible to pass parameters via the transfer so the XML would look like this: There was a hint about using something called "inline" which indicated that an XML dialplan would not be needed. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/JavaScript-control-Transfer-tp5160890p5161043.html Sent from the freeswitch-users mailing list archive at Nabble.com. From lloyd.aloysius at gmail.com Wed Jun 9 22:26:51 2010 From: lloyd.aloysius at gmail.com (Aloysius Lloyd) Date: Thu, 10 Jun 2010 01:26:51 -0400 Subject: [Freeswitch-users] LUA Callback - Help Message-ID: Hi All, I am trying to build a callback application using LUA. I have the following setup. - Dial the DID number - Hangup - Callback lua initiate the call after 10 secods The above working great. But when I answer the call ...the wav file never play on the channel.What I am doing wrong? *callback.lua* os.execute("sleep " .. tonumber(10)); pin_number_test="0511"; out_gateway1="sofia/gateway/voipms/1"; message_dir="/usr/local/freeswitch/sounds/en/us/callie/custom/callback/"; number_to_call= argv[1]; session = freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); session:setAutoHangup(false); if ( session:ready() ) then session:sleep(3000); session:streamFile(message_dir.."enter-dest.wav"); end Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/69c56e33/attachment.html From lloyd.aloysius at gmail.com Wed Jun 9 22:42:57 2010 From: lloyd.aloysius at gmail.com (Aloysius Lloyd) Date: Thu, 10 Jun 2010 01:42:57 -0400 Subject: [Freeswitch-users] LUA Callback - Help In-Reply-To: References: Message-ID: I did a simple test with javascript. The same setup working with the following javascript. session = new Session('sofia/gateway/voipms/14161231234'); session.waitForAnswer(10000); if (session.ready()) { session.sleep(1000); session.streamFile('/usr/local/freeswitch/sounds/en/us/callie/custom/callback/enter-dest.wav'); } --- Thanks Lloyd On Thu, Jun 10, 2010 at 1:26 AM, Aloysius Lloyd wrote: > Hi All, > > I am trying to build a callback application using LUA. I have the following > setup. > > - Dial the DID number > - Hangup > - Callback lua initiate the call after 10 secods > > The above working great. > > But when I answer the call ...the wav file never play on the channel.What I > am doing wrong? > > > > > > > > > > *callback.lua* > > os.execute("sleep " .. tonumber(10)); > > pin_number_test="0511"; > out_gateway1="sofia/gateway/voipms/1"; > message_dir="/usr/local/freeswitch/sounds/en/us/callie/custom/callback/"; > > number_to_call= argv[1]; > > session = > freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); > session:setAutoHangup(false); > > if ( session:ready() ) then > session:sleep(3000); > session:streamFile(message_dir.."enter-dest.wav"); > end > > Thanks > Lloyd > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/28713e7c/attachment.html From sid at eltc.ru Wed Jun 9 23:00:42 2010 From: sid at eltc.ru (Sergey Scheglov) Date: Thu, 10 Jun 2010 13:00:42 +0700 Subject: [Freeswitch-users] LUA Callback - Help In-Reply-To: References: Message-ID: <20100610130042.4802f558@shadow.elt> ? Thu, 10 Jun 2010 01:26:51 -0400 Aloysius Lloyd wrote: > session = > freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); > session:setAutoHangup(false); > > Thanks > Lloyd Try without bypass_media_after_bridge=true Regard Sergey Scheglov From lloyd.aloysius at sunteltech.ca Wed Jun 9 23:07:20 2010 From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd) Date: Thu, 10 Jun 2010 02:07:20 -0400 Subject: [Freeswitch-users] LUA Callback - Help In-Reply-To: <20100610130042.4802f558@shadow.elt> References: <20100610130042.4802f558@shadow.elt> Message-ID: I try the bypass_media_after_bridge=true .. same results. Thanks Lloyd On Thu, Jun 10, 2010 at 2:00 AM, Sergey Scheglov wrote: > ? Thu, 10 Jun 2010 01:26:51 -0400 > Aloysius Lloyd wrote: > > > > session = > > > freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); > > session:setAutoHangup(false); > > > > Thanks > > Lloyd > > Try without bypass_media_after_bridge=true > > Regard > Sergey Scheglov > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/7278e58b/attachment-0001.html From lloyd.aloysius at sunteltech.ca Wed Jun 9 23:07:48 2010 From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd) Date: Thu, 10 Jun 2010 02:07:48 -0400 Subject: [Freeswitch-users] LUA Callback - Help In-Reply-To: References: <20100610130042.4802f558@shadow.elt> Message-ID: without bypass_media_after_bridge=true same results. Lloyd 2010/6/10 Aloysius Lloyd > I try the bypass_media_after_bridge=true .. same results. > > Thanks > Lloyd > > > > > On Thu, Jun 10, 2010 at 2:00 AM, Sergey Scheglov wrote: > >> ? Thu, 10 Jun 2010 01:26:51 -0400 >> Aloysius Lloyd wrote: >> >> >> > session = >> > >> freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); >> > session:setAutoHangup(false); >> > >> > Thanks >> > Lloyd >> >> Try without bypass_media_after_bridge=true >> >> Regard >> Sergey Scheglov >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/ca7858c0/attachment.html From infos at madovsky.org Wed Jun 9 23:16:48 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 02:16:48 -0400 Subject: [Freeswitch-users] mod_lcr Message-ID: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?ThanksFranck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/ad7f45fb/attachment.html From gcd at i.ph Wed Jun 9 23:44:01 2010 From: gcd at i.ph (Nandy Dagondon) Date: Thu, 10 Jun 2010 14:44:01 +0800 Subject: [Freeswitch-users] mod_lcr In-Reply-To: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> Message-ID: i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: > I'm experimenting with mod_lcr with postgresql (8.4.4) > there s an example of custom sql on wiki below : > > > > > > > however the query failed cause of digits_prefix field doesn't exist in the table. > > is it a typo ? or does it need a field concatenation of prefix and digits ? > > Thanks > > > > Franck > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/8b644b0d/attachment.html From infos at madovsky.org Wed Jun 9 23:56:26 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 02:56:26 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> Message-ID: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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/20100610/1a897301/attachment.html From gcd at i.ph Thu Jun 10 00:09:15 2010 From: gcd at i.ph (Nandy Dagondon) Date: Thu, 10 Jun 2010 15:09:15 +0800 Subject: [Freeswitch-users] mod_lcr In-Reply-To: References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> Message-ID: it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: > ok so it needs also the alias l.digits in the condition I think. > I'm a little confused about digits and prefix. > if I check a number with the country code is it need to join > prefix+digits ? how with this kinkd of sql request ? > > Thanks > > F > > ----- Original Message ----- > *From:* Nandy Dagondon > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, June 10, 2010 2:44 AM > *Subject:* Re: [Freeswitch-users] mod_lcr > > i think it's a typo. i changed digits_prefix to digits. to be sure, pls > check the CREATE TABLE entries. > -nandy > > > On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: > >> I'm experimenting with mod_lcr with postgresql (8.4.4) >> there s an example of custom sql on wiki below : >> >> >> >> >> >> >> however the query failed cause of digits_prefix field doesn't exist in the table. >> >> is it a typo ? or does it need a field concatenation of prefix and digits ? >> >> Thanks >> >> >> >> Franck >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/a2cbb07c/attachment-0001.html From shaikbashaatc at yahoo.com Thu Jun 10 00:53:16 2010 From: shaikbashaatc at yahoo.com (Shaik basha) Date: Thu, 10 Jun 2010 00:53:16 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch download link and configuration Setup webpage on website Message-ID: <113855.18830.qm@web43407.mail.sp1.yahoo.com> Is there anyone who can help me in this forum? As I have not seen any reply to my earlier mails. Your earliest response in this regard would be very much appreciated. Thanking you in advance. REgards, > Good morning every one. I a m a new bie in freeswitch, > though I tried to search the download page and configuration > set up. but, I don't see any where. Can any one help me in > this regard. I have spent several hours, though I was not > succeeded. > > Hence I kindly request to let me know from where I can > download and how to do configuration setup. Thanking in > advance. earliest response in this regard would be very much > appreciated. I would be very thankful and grateful for your > kind information. Regards, > > shaikbashaatc > +919246769086 > > > ? ? ? > From infos at madovsky.org Thu Jun 10 00:53:57 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 03:53:57 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> Message-ID: <763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> ok thanks I will read again F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 3:09 AM Subject: Re: [Freeswitch-users] mod_lcr it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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/20100610/ff1fcc0c/attachment.html From daniel.neubert at solomo.de Thu Jun 10 01:27:53 2010 From: daniel.neubert at solomo.de (Daniel Neubert) Date: Thu, 10 Jun 2010 10:27:53 +0200 Subject: [Freeswitch-users] Freeswitch download link and configuration Setup webpage on website In-Reply-To: <113855.18830.qm@web43407.mail.sp1.yahoo.com> References: <113855.18830.qm@web43407.mail.sp1.yahoo.com> Message-ID: <4C10A209.6050802@solomo.de> Have you read the wiki articles? http://wiki.freeswitch.org/wiki/Main_Page You'll find very nice instructions how to download, compile and install freeswitch. Best regards / Mit freundlichen Gr??en, Daniel Neubert On 10.06.2010 09:53, Shaik basha wrote: > Is there anyone who can help me in this forum? As I have not seen any reply to my earlier mails. Your earliest response in this regard would be very much appreciated. Thanking you in advance. REgards, > > > >> Good morning every one. I a m a new bie in freeswitch, >> though I tried to search the download page and configuration >> set up. but, I don't see any where. Can any one help me in >> this regard. I have spent several hours, though I was not >> succeeded. >> >> Hence I kindly request to let me know from where I can >> download and how to do configuration setup. Thanking in >> advance. earliest response in this regard would be very much >> appreciated. I would be very thankful and grateful for your >> kind information. Regards, >> >> shaikbashaatc >> +919246769086 >> >> >> >> >> > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/a772aba6/attachment.html From gcd at i.ph Thu Jun 10 01:32:34 2010 From: gcd at i.ph (Nandy Dagondon) Date: Thu, 10 Jun 2010 16:32:34 +0800 Subject: [Freeswitch-users] Freeswitch download link and configuration Setup webpage on website In-Reply-To: <113855.18830.qm@web43407.mail.sp1.yahoo.com> References: <113855.18830.qm@web43407.mail.sp1.yahoo.com> Message-ID: it's here http://wiki.freeswitch.org/wiki/Installation_Guide -nandy On Thu, Jun 10, 2010 at 3:53 PM, Shaik basha wrote: > Is there anyone who can help me in this forum? As I have not seen any reply > to my earlier mails. Your earliest response in this regard would be very > much appreciated. Thanking you in advance. REgards, > > > > Good morning every one. I a m a new bie in freeswitch, > > though I tried to search the download page and configuration > > set up. but, I don't see any where. Can any one help me in > > this regard. I have spent several hours, though I was not > > succeeded. > > > > Hence I kindly request to let me know from where I can > > download and how to do configuration setup. Thanking in > > advance. earliest response in this regard would be very much > > appreciated. I would be very thankful and grateful for your > > kind information. Regards, > > > > shaikbashaatc > > +919246769086 > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/b2aa77cc/attachment.html From azatek0 at gmail.com Thu Jun 10 01:55:32 2010 From: azatek0 at gmail.com (Aza Tek) Date: Thu, 10 Jun 2010 10:55:32 +0200 Subject: [Freeswitch-users] Freeswitch in an existing phone network In-Reply-To: References: <4C0E3FCE.1040803@iinet.net.au> <4C0FA8AD.8020907@communicatefreely.net> Message-ID: Whilst on Git, is possible to only 'checkout' certain components of FreeSWITCH as submodules using Git? Thanks Aza On Thu, Jun 10, 2010 at 1:40 AM, Michael Collins wrote: > > > It will help you get up and running quickly. (Note: we switched to git from > svn, so we recommend you use git to download the source...) > > Have fun! > -MC > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/19497a62/attachment-0001.html From freeswitch at peely.com Thu Jun 10 02:34:56 2010 From: freeswitch at peely.com (peely) Date: Thu, 10 Jun 2010 02:34:56 -0700 (PDT) Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> <1275859376110-5146705.post@n2.nabble.com> <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> <1275862175540-5146803.post@n2.nabble.com> <1276001266045-5153498.post@n2.nabble.com> Message-ID: <1276162496035-5162317.post@n2.nabble.com> Hi, It's not a hijacked thread, I created this with an issue around uuid_media, initially it was hanging, then in the latest git it receives a 4xx from the far end due to the Requires: timer support bot being supported in the reinvite transaction. I did anecdotally mention problems with events, which I subsequently pulled off to another thread. I was looking at RFC 4028, in the examples it shows the refresh event not providing the Required: timer header even for the refresh of the original invite transaction, just the Supported: timer. There's nothing in RFC 3725 regarding sesison timers in 3PCC but as the reinvite for the media adjustment is a new transaction, I don't think it needs the Requires: timer option to persist the timer support on the original dialogue. This is certainly the behaviour of the previous equipment I had on this interconnect and it operates using session timers and 3PCC in several other environments too. I'm not brilliant with C, but if you could please point me to the right approach to removing the Requires: timer field in the header I'd be happy to test it and let you know if it causes any problems with rienvites across various carriers and other platforms. Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5162317.html Sent from the freeswitch-users mailing list archive at Nabble.com. From azatek0 at gmail.com Thu Jun 10 03:27:01 2010 From: azatek0 at gmail.com (Aza Tek) Date: Thu, 10 Jun 2010 12:27:01 +0200 Subject: [Freeswitch-users] Weekly FreeSwitch Conference call June 9 @ 1pm EST In-Reply-To: References: Message-ID: It seems a lot of the Weekly Conference Calls actually don't have associated recordings, is this correct or am I just missing them? I've seen some recordings in the Wiki and Conf Call pages, but for less than half of all the conferences that are there. Should I also be looking somewhere else? Thanks Aza > On Tue, Jun 8, 2010 at 11:22 AM, Norman Tomlins wrote: > >> Weekly FreeSwitch Conference call June 9 @ 1pm EST >> >> >> >> Hi Everyone, >> >> >> >> I would just like to remind everyone that we have our weekly conference >> call on June , 9th @ 1pm EST. >> >> >> During this conference call Darren Schreiber will talk about >> mod_nibblebill. >> >> >> Information on mod_nilbble can be found here: >> http://wiki.freeswitch.org/wiki/Mod_nibblebill >> >> Information on Weekly Conference call: >> http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_09 >> >> >> >> Norman Tomlins >> >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/73892269/attachment.html From vetali100 at gmail.com Thu Jun 10 04:17:51 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Thu, 10 Jun 2010 14:17:51 +0300 Subject: [Freeswitch-users] LUA Callback - Help In-Reply-To: References: <20100610130042.4802f558@shadow.elt> Message-ID: It looks OK what you wrote - we have almost same lua code. The only thing that I have in addition in my script is "if (session:answered())" construction. Try to write it like this: ... if ( session:ready() ) then session:sleep(1000); if (session:answered()) then session:sleep(1000); session:streamFile(message_dir.."enter-dest.wav"); end end ... Not sure why this should work and that does not, but this is how it works for me - worth trying at lest. Regards, Vitalie 10 ???? 2010 ?. 9:07 ???????????? Aloysius Lloyd < lloyd.aloysius at sunteltech.ca> ???????: > > without bypass_media_after_bridge=true same results. > > Lloyd > > > > 2010/6/10 Aloysius Lloyd > > I try the bypass_media_after_bridge=true .. same results. >> >> Thanks >> Lloyd >> >> >> >> >> On Thu, Jun 10, 2010 at 2:00 AM, Sergey Scheglov wrote: >> >>> ? Thu, 10 Jun 2010 01:26:51 -0400 >>> Aloysius Lloyd wrote: >>> >>> >>> > session = >>> > >>> freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); >>> > session:setAutoHangup(false); >>> > >>> > Thanks >>> > Lloyd >>> >>> Try without bypass_media_after_bridge=true >>> >>> Regard >>> Sergey Scheglov >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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/20100610/a465ccad/attachment.html From rupa at rupa.com Thu Jun 10 05:38:42 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 10 Jun 2010 07:38:42 -0500 Subject: [Freeswitch-users] mod_lcr In-Reply-To: <763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> <763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: Read the part about how to define the table + the gist index. The whole custom_sql thing assumes a familiarity with sql. you can choose to not have a digits_prefix column and just change the datatype of prefix to prefix. You can do what I did which is to have prefix be text and digits_prefix be of type prefix and a trigger to keep the two in sync. the key is that you are searching against the prefix column for which there is a GIST index. On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: > ok thanks I will read again > > F > > ----- Original Message ----- > *From:* Nandy Dagondon > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, June 10, 2010 3:09 AM > *Subject:* Re: [Freeswitch-users] mod_lcr > > it's the digits_prefix in the WHERE clause that's causing the error. > > ur question re prefix+digits, it's explained in the Custom SQL portion in > the wiki. > > -nandy > > > On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: > >> ok so it needs also the alias l.digits in the condition I think. >> I'm a little confused about digits and prefix. >> if I check a number with the country code is it need to join >> prefix+digits ? how with this kinkd of sql request ? >> >> Thanks >> >> F >> >> ----- Original Message ----- >> *From:* Nandy Dagondon >> *To:* freeswitch-users at lists.freeswitch.org >> *Sent:* Thursday, June 10, 2010 2:44 AM >> *Subject:* Re: [Freeswitch-users] mod_lcr >> >> i think it's a typo. i changed digits_prefix to digits. to be sure, pls >> check the CREATE TABLE entries. >> -nandy >> >> >> On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: >> >>> I'm experimenting with mod_lcr with postgresql (8.4.4) >>> there s an example of custom sql on wiki below : >>> >>> >>> >>> >>> >>> >>> however the query failed cause of digits_prefix field doesn't exist in the table. >>> >>> is it a typo ? or does it need a field concatenation of prefix and digits ? >>> >>> Thanks >>> >>> >>> >>> Franck >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/9f576dd4/attachment-0001.html From oliver.schenk at iinet.net.au Thu Jun 10 05:48:31 2010 From: oliver.schenk at iinet.net.au (Oliver Schenk) Date: Thu, 10 Jun 2010 20:48:31 +0800 Subject: [Freeswitch-users] Freeswitch in an existing phone network In-Reply-To: <4C0FA8AD.8020907@communicatefreely.net> References: <4C0E3FCE.1040803@iinet.net.au> <4C0FA8AD.8020907@communicatefreely.net> Message-ID: <4C10DF1F.5020802@iinet.net.au> Hi, Thanks for everyone's reply. Great help. I have a basic IVR going when I dial a given extension. I have managed to clean out most of FreeSWITCH config and I've even written a little javascript file to which my call gets redirected to and it just says a small phrase using flite. I'm looking at buying more professional TTS voices such as Cebstral or Ivona. I'm hosting my freeswitch server in a Virtual Box instance running on Ubuntu Server 10.04. I do my testing from X-Lite running on the host computer. Anyway, that said, I think we will have to stick with Windows because: 1) Our existing railway SCADA server will be windows based. 2) The SCADA libraries are .NET based. 3) I don't want to have separate hardware just to run linux. 4) Our company isn't too good at supporting linux. My problem now is to try to somehow trigger freeswitch to make an outgoing call on demand... The flow of my logic is: 1) A critical alarm occurs on our SCADA system. 2) FreeSWITCH should be triggered to make an outgoing call. 3) Use a pre-recorded WAV or use TTS to tell the callee what the alarm is. The problem: I don't know how to "tell" freeswitch that it needs to make a call and maybe even pass some parameters. Given I have C# and SQL at my disposal, what should I use? A freeswitch javascript program that runs in an infinite loop and constantly scans for new alarms in a database? In the registry? In a folder using files? Using a http request? command line? In other words, how do I trigger a freeswitch javascript program externally? (By the way I tried to use the "managed" module that comes with freeswitch, but it just throws exceptions left right and center TypeInitializationException)... Anyone have some experience? Thanks, Oliver Schenk On 9/06/2010 10:43 PM, Tim St. Pierre wrote: > Hi Oliver, > > FreeSwitch will do whatever you tell it to do and no more. > > Here's a few suggestions though - > > Empty out the default dialplan directory. Don't throw those away, as you may want to reference them > as examples, but move them somewhere else. > > Edit modules.conf.xml and comment or remove any modules that you don't need. This will also save > memory and other resources. > > You can also disable all the SIP profiles except one. Pick one that makes sense (either Internal or > External, it doesn't really matter that much), and edit it so that it makes sense with respect to > your network. What is your topology? Will you just be setting freeswitch up with a static IP > address and having calls sent to it by the main PBX? If that's the case, you can disable a lot of > the STUN and uPNP functionality. Tell this profile to bind to the IP and port that the PBX will > send the calls to. > > Then all you have to do, is create a very simple dialplan that will answer an incoming call and > perform whatever task you want. You would essentially be starting with a blank sheet, adding just > the functions that you want. > > Hope that makes sense. > > -Tim > > Oliver Schenk wrote: > >> Hi All, >> >> The company I work for currently has quite an extensive phone network >> which gets carried between old analogue PABXes which also has an >> interface to IP based phones. All the phones in our office are connected >> via CAT5 cable using IP, however literally hundreds of phones out in the >> field (we operate railway infrastructure) are on standard voice analogue >> phones carried through fibreoptics. >> >> Anyway, I would like to use Freeswitch purely for its IVR and TTS >> abilities and nothing else. So basically I just need it to act like a >> slave to whatever IP phone network is already out there. I'm a bit >> worried if I fire up freeswitch it will hijack the phone network! >> >> All our phones are accessible via a 5 digit extension. I would like >> Freeswitch to be behind one of those ... say 12345. If anyone within our >> phone network dials 12345 then Freeswitch should answer. I guess my >> question is...how should I go about disabling most of FreeSwitch except >> it's ability to pick up the phone and speak IVR/TTS and make an outgoing >> call via the existing phone network? >> >> Any general pointers appreciated. >> >> >> Thanks, >> >> Oliver Schenk >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > From david.ponzone at gmail.com Thu Jun 10 06:05:29 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 10 Jun 2010 15:05:29 +0200 Subject: [Freeswitch-users] Freeswitch in an existing phone network In-Reply-To: <4C10DF1F.5020802@iinet.net.au> References: <4C0E3FCE.1040803@iinet.net.au> <4C0FA8AD.8020907@communicatefreely.net> <4C10DF1F.5020802@iinet.net.au> Message-ID: You will have to use ESL. Check the wiki. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 10/06/2010 ? 14:48, Oliver Schenk a ?crit : > Hi, > > Thanks for everyone's reply. Great help. I have a basic IVR going > when I > dial a given extension. > > I have managed to clean out most of FreeSWITCH config and I've even > written a little javascript file to which my call gets redirected to > and > it just says a small phrase using flite. I'm looking at buying more > professional TTS voices such as Cebstral or Ivona. I'm hosting my > freeswitch server in a Virtual Box instance running on Ubuntu Server > 10.04. I do my testing from X-Lite running on the host computer. > > Anyway, that said, I think we will have to stick with Windows because: > 1) Our existing railway SCADA server will be windows based. > 2) The SCADA libraries are .NET based. > 3) I don't want to have separate hardware just to run linux. > 4) Our company isn't too good at supporting linux. > > > My problem now is to try to somehow trigger freeswitch to make an > outgoing call on demand... > > > The flow of my logic is: > > 1) A critical alarm occurs on our SCADA system. > 2) FreeSWITCH should be triggered to make an outgoing call. > 3) Use a pre-recorded WAV or use TTS to tell the callee what the > alarm is. > > > The problem: > > I don't know how to "tell" freeswitch that it needs to make a call and > maybe even pass some parameters. Given I have C# and SQL at my > disposal, > what should I use? A freeswitch javascript program that runs in an > infinite loop and constantly scans for new alarms in a database? In > the > registry? In a folder using files? Using a http request? command line? > > In other words, how do I trigger a freeswitch javascript program > externally? > > > (By the way I tried to use the "managed" module that comes with > freeswitch, but it just throws exceptions left right and center > TypeInitializationException)... > > Anyone have some experience? > > > Thanks, > > Oliver Schenk > > > On 9/06/2010 10:43 PM, Tim St. Pierre wrote: >> Hi Oliver, >> >> FreeSwitch will do whatever you tell it to do and no more. >> >> Here's a few suggestions though - >> >> Empty out the default dialplan directory. Don't throw those away, >> as you may want to reference them >> as examples, but move them somewhere else. >> >> Edit modules.conf.xml and comment or remove any modules that you >> don't need. This will also save >> memory and other resources. >> >> You can also disable all the SIP profiles except one. Pick one >> that makes sense (either Internal or >> External, it doesn't really matter that much), and edit it so that >> it makes sense with respect to >> your network. What is your topology? Will you just be setting >> freeswitch up with a static IP >> address and having calls sent to it by the main PBX? If that's the >> case, you can disable a lot of >> the STUN and uPNP functionality. Tell this profile to bind to the >> IP and port that the PBX will >> send the calls to. >> >> Then all you have to do, is create a very simple dialplan that will >> answer an incoming call and >> perform whatever task you want. You would essentially be starting >> with a blank sheet, adding just >> the functions that you want. >> >> Hope that makes sense. >> >> -Tim >> >> Oliver Schenk wrote: >> >>> Hi All, >>> >>> The company I work for currently has quite an extensive phone >>> network >>> which gets carried between old analogue PABXes which also has an >>> interface to IP based phones. All the phones in our office are >>> connected >>> via CAT5 cable using IP, however literally hundreds of phones out >>> in the >>> field (we operate railway infrastructure) are on standard voice >>> analogue >>> phones carried through fibreoptics. >>> >>> Anyway, I would like to use Freeswitch purely for its IVR and TTS >>> abilities and nothing else. So basically I just need it to act >>> like a >>> slave to whatever IP phone network is already out there. I'm a bit >>> worried if I fire up freeswitch it will hijack the phone network! >>> >>> All our phones are accessible via a 5 digit extension. I would like >>> Freeswitch to be behind one of those ... say 12345. If anyone >>> within our >>> phone network dials 12345 then Freeswitch should answer. I guess my >>> question is...how should I go about disabling most of FreeSwitch >>> except >>> it's ability to pick up the phone and speak IVR/TTS and make an >>> outgoing >>> call via the existing phone network? >>> >>> Any general pointers appreciated. >>> >>> >>> Thanks, >>> >>> Oliver Schenk >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/a6e4468a/attachment-0001.html From macedoslm at gmail.com Thu Jun 10 06:16:41 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Thu, 10 Jun 2010 10:16:41 -0300 Subject: [Freeswitch-users] Gateway Registration In-Reply-To: References: Message-ID: Thank you for the responses. You solved my problem. Regards, -- Samuel Macedo On 9 June 2010 20:34, leonardo alves wrote: > No you register 2 gateways. > > > > > > > > > > > > > value="YOUR SIP user name2"/> name="from-domain" value="sip.domain.com"/> value="3600"/> > > And in the dial plan you can do the bridge with the gateway you want using > the name of the gateway. > Leonardo > On Wed, Jun 9, 2010 at 5:19 PM, Samuel Macedo wrote: > >> How can I set the gateway domain? Can it be different from the gateway >> name? >> Example: >> >> >> >> >> >> >> >> >> >> *Gateway Name*: gateway.com >> *Gateway Domain*: gateway.com >> >> Thanks, >> -- >> Samuel Macedo >> >> >> On 8 June 2010 13:03, Samuel Macedo wrote: >> >>> But the name of the gateway is the domain that will be used to register, >>> or no? >>> >>> -- >>> Samuel Macedo >>> >>> >>> On 8 June 2010 06:46, Steven Ayre wrote: >>> >>>> Configure the same gateway twice, with different names. Which gateway >>>> name you use will determine which username is used. >>>> >>>> >>>> >>>> On 7 June 2010 20:26, Samuel Macedo wrote: >>>> >>>>> Hi, >>>>> >>>>> I want to use the same Gateway but with different users and passwords >>>>> to make and receive calls. >>>>> This is the scenario: >>>>> I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch >>>>> UserAgents (Sofia External) to this gateway so I can receive inbound calls. >>>>> And to make a Outbound call I want to decide witch UserAgent I will use. >>>>> >>>>> Is there anyway to do this? >>>>> >>>>> Regards, >>>>> -- >>>>> Samuel Macedo >>>>> Belo Horizonte - Brazil >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.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 >>>> >>>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100610/6fcf22fb/attachment.html From oliver.schenk at iinet.net.au Thu Jun 10 06:19:35 2010 From: oliver.schenk at iinet.net.au (Oliver Schenk) Date: Thu, 10 Jun 2010 21:19:35 +0800 Subject: [Freeswitch-users] Freeswitch in an existing phone network In-Reply-To: <4C10DF1F.5020802@iinet.net.au> References: <4C0E3FCE.1040803@iinet.net.au> <4C0FA8AD.8020907@communicatefreely.net> <4C10DF1F.5020802@iinet.net.au> Message-ID: <4C10E667.8030001@iinet.net.au> Just some more info regarding managed C# and ESL. This is what I did: Put the following line in modules.conf.xml: I also checked that the module is in the mod folder. I then started Freeswitch from cmd in windows using "Freeswitch.exe -nonat". There are no errors while it boots. I do however get all sorts of strange IP addresses. My own LAN is a 10.1.1.* network, but I also get IP addresses like: 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12 (deny) [] to list wan.auto 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16 (deny) [] to list wan.auto 2010-06-10 21:00:05.530374 [NOTICE] switch_core.c:970 Created ip list nat.auto default (deny) 2010-06-10 21:00:05.530374 [NOTICE] switch_core.c:972 Adding 10.1.1.4/255.255.255.0 (deny) to list nat.auto 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8 (allow) [] to list nat.auto 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding 172.16.0.0/12 (allow) [] to list nat.auto 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding 192.168.0.0/16 (allow) [] to list nat.auto 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:981 Created ip list loopback.auto default (deny) 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding 127.0.0.0/8 (allow) [] to list loopback.auto 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:987 Created ip list localnet.auto default (deny) 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:990 Adding 10.1.1.4/255.255.255.0 (allow) to list localnet.auto 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding 192.168.42.0/24 (deny) [] to list lan 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:1086 Adding 192.168.42.0/24 (deny) to list lan 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding 192.168.42.42/32 (allow) [] to list lan 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:1086 Adding 192.168.42.42/32 (allow) to list lan 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding 192.0.2.0/24 (allow) [brian at templar.ath.cx] to list domains Where do all those IP addresses come from? Why are they there? Ok, next thing I did was open up VS2008 and opened up freeswitch-1.0.6/libs/esl/managed/managed_esl.2008.sln. No problems there. I then hit compile and it worked without any errors. I had the following uncommented: ThreadPool.QueueUserWorkItem(new WaitCallback(InboundMode)); //ThreadPool.QueueUserWorkItem(new WaitCallback(OutboundModeSync)); //ThreadPool.QueueUserWorkItem(new WaitCallback(OutboundModeAsync)); I then hit run and got the following error: An unhandled exception of type 'System.TypeInitializationException' occurred in ManagedEsl.dll Additional information: The type initializer for 'ESLPINVOKE' threw an exception. It breaks here: public ESLconnection(string host, string port, string password) : this(ESLPINVOKE.new_ESLconnection__SWIG_1(host, port, password), true) { } Output: A first chance exception of type 'System.BadImageFormatException' occurred in ManagedEsl.dll A first chance exception of type 'System.TypeInitializationException' occurred in ManagedEsl.dll A first chance exception of type 'System.TypeInitializationException' occurred in ManagedEsl.dll An unhandled exception of type 'System.TypeInitializationException' occurred in ManagedEsl.dll Additional information: The type initializer for 'ESLPINVOKE' threw an exception. Unfortunately I have no idea how to fix it! Thanks, Oliver Schenk On 10/06/2010 8:48 PM, Oliver Schenk wrote: > Hi, > > Thanks for everyone's reply. Great help. I have a basic IVR going when I > dial a given extension. > > I have managed to clean out most of FreeSWITCH config and I've even > written a little javascript file to which my call gets redirected to and > it just says a small phrase using flite. I'm looking at buying more > professional TTS voices such as Cebstral or Ivona. I'm hosting my > freeswitch server in a Virtual Box instance running on Ubuntu Server > 10.04. I do my testing from X-Lite running on the host computer. > > Anyway, that said, I think we will have to stick with Windows because: > 1) Our existing railway SCADA server will be windows based. > 2) The SCADA libraries are .NET based. > 3) I don't want to have separate hardware just to run linux. > 4) Our company isn't too good at supporting linux. > > > My problem now is to try to somehow trigger freeswitch to make an > outgoing call on demand... > > > The flow of my logic is: > > 1) A critical alarm occurs on our SCADA system. > 2) FreeSWITCH should be triggered to make an outgoing call. > 3) Use a pre-recorded WAV or use TTS to tell the callee what the alarm is. > > > The problem: > > I don't know how to "tell" freeswitch that it needs to make a call and > maybe even pass some parameters. Given I have C# and SQL at my disposal, > what should I use? A freeswitch javascript program that runs in an > infinite loop and constantly scans for new alarms in a database? In the > registry? In a folder using files? Using a http request? command line? > > In other words, how do I trigger a freeswitch javascript program externally? > > > (By the way I tried to use the "managed" module that comes with > freeswitch, but it just throws exceptions left right and center > TypeInitializationException)... > > Anyone have some experience? > > > Thanks, > > Oliver Schenk > > > On 9/06/2010 10:43 PM, Tim St. Pierre wrote: > >> Hi Oliver, >> >> FreeSwitch will do whatever you tell it to do and no more. >> >> Here's a few suggestions though - >> >> Empty out the default dialplan directory. Don't throw those away, as you may want to reference them >> as examples, but move them somewhere else. >> >> Edit modules.conf.xml and comment or remove any modules that you don't need. This will also save >> memory and other resources. >> >> You can also disable all the SIP profiles except one. Pick one that makes sense (either Internal or >> External, it doesn't really matter that much), and edit it so that it makes sense with respect to >> your network. What is your topology? Will you just be setting freeswitch up with a static IP >> address and having calls sent to it by the main PBX? If that's the case, you can disable a lot of >> the STUN and uPNP functionality. Tell this profile to bind to the IP and port that the PBX will >> send the calls to. >> >> Then all you have to do, is create a very simple dialplan that will answer an incoming call and >> perform whatever task you want. You would essentially be starting with a blank sheet, adding just >> the functions that you want. >> >> Hope that makes sense. >> >> -Tim >> >> Oliver Schenk wrote: >> >> >>> Hi All, >>> >>> The company I work for currently has quite an extensive phone network >>> which gets carried between old analogue PABXes which also has an >>> interface to IP based phones. All the phones in our office are connected >>> via CAT5 cable using IP, however literally hundreds of phones out in the >>> field (we operate railway infrastructure) are on standard voice analogue >>> phones carried through fibreoptics. >>> >>> Anyway, I would like to use Freeswitch purely for its IVR and TTS >>> abilities and nothing else. So basically I just need it to act like a >>> slave to whatever IP phone network is already out there. I'm a bit >>> worried if I fire up freeswitch it will hijack the phone network! >>> >>> All our phones are accessible via a 5 digit extension. I would like >>> Freeswitch to be behind one of those ... say 12345. If anyone within our >>> phone network dials 12345 then Freeswitch should answer. I guess my >>> question is...how should I go about disabling most of FreeSwitch except >>> it's ability to pick up the phone and speak IVR/TTS and make an outgoing >>> call via the existing phone network? >>> >>> Any general pointers appreciated. >>> >>> >>> Thanks, >>> >>> Oliver Schenk >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 lloyd.aloysius at sunteltech.ca Thu Jun 10 06:20:45 2010 From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd) Date: Thu, 10 Jun 2010 09:20:45 -0400 Subject: [Freeswitch-users] LUA Callback - Help In-Reply-To: References: <20100610130042.4802f558@shadow.elt> Message-ID: VItalli, I try your suggestion. Now the script look like this os.execute("sleep " .. tonumber(10)); pin_number_test="0511"; out_gateway1="sofia/gateway/voipms/1"; message_dir="/usr/local/freeswitch/sounds/en/us/callie/custom/callback/"; number_to_call= argv[1]; session = freeswitch.Session("{ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); session:setAutoHangup(false); if ( session:ready() ) then session:sleep(1000); if (session:answered()) then session:sleep(1000); session:streamFile(message_dir.."enter-dest.wav"); end end === Also I try 5 test . This time out of 5 test, wave file play only one time. Other for 4 tests silent. Thanks Lloyd 2010/6/10 Vitalii Colosov > It looks OK what you wrote - we have almost same lua code. > > The only thing that I have in addition in my script is "if > (session:answered())" construction. > > Try to write it like this: > ... > > if ( session:ready() ) then > session:sleep(1000); > if (session:answered()) then > session:sleep(1000); > session:streamFile(message_dir.."enter-dest.wav"); > end > > end > ... > > Not sure why this should work and that does not, but this is how it works > for me - worth trying at lest. > > > Regards, > Vitalie > > > 10 ???? 2010 ?. 9:07 ???????????? Aloysius Lloyd < > lloyd.aloysius at sunteltech.ca> ???????: > > >> without bypass_media_after_bridge=true same results. >> >> Lloyd >> >> >> >> 2010/6/10 Aloysius Lloyd >> >> I try the bypass_media_after_bridge=true .. same results. >>> >>> Thanks >>> Lloyd >>> >>> >>> >>> >>> On Thu, Jun 10, 2010 at 2:00 AM, Sergey Scheglov wrote: >>> >>>> ? Thu, 10 Jun 2010 01:26:51 -0400 >>>> Aloysius Lloyd wrote: >>>> >>>> >>>> > session = >>>> > >>>> freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); >>>> > session:setAutoHangup(false); >>>> > >>>> > Thanks >>>> > Lloyd >>>> >>>> Try without bypass_media_after_bridge=true >>>> >>>> Regard >>>> Sergey Scheglov >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/8347c541/attachment-0001.html From david.ponzone at gmail.com Thu Jun 10 06:29:46 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 10 Jun 2010 15:29:46 +0200 Subject: [Freeswitch-users] Freeswitch in an existing phone network In-Reply-To: <4C10E667.8030001@iinet.net.au> References: <4C0E3FCE.1040803@iinet.net.au> <4C0FA8AD.8020907@communicatefreely.net> <4C10DF1F.5020802@iinet.net.au> <4C10E667.8030001@iinet.net.au> Message-ID: <74EA08EE-7A7A-4E83-90F6-641ED1D6BF65@gmail.com> Oliver, those IPs, as you probably noticed, are all RFC1918. By default, FS will do some automatic NAT detection when a device's IP is a private one. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 10/06/2010 ? 15:19, Oliver Schenk a ?crit : > Just some more info regarding managed C# and ESL. This is what I did: > > Put the following line in modules.conf.xml: > > > > I also checked that the module is in the mod folder. > > > > I then started Freeswitch from cmd in windows using "Freeswitch.exe - > nonat". > > There are no errors while it boots. I do however get all sorts of > strange IP addresses. My own LAN is a 10.1.1.* network, but I also get > IP addresses like: > > 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding > 172.16.0.0/12 (deny) [] to list wan.auto > 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding > 192.168.0.0/16 (deny) [] to list wan.auto > 2010-06-10 21:00:05.530374 [NOTICE] switch_core.c:970 Created ip list > nat.auto default (deny) > 2010-06-10 21:00:05.530374 [NOTICE] switch_core.c:972 Adding > 10.1.1.4/255.255.255.0 (deny) to list nat.auto > 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding > 10.0.0.0/8 > (allow) [] to list nat.auto > 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding > 172.16.0.0/12 (allow) [] to list nat.auto > 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding > 192.168.0.0/16 (allow) [] to list nat.auto > 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:981 Created ip list > loopback.auto default (deny) > 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding > 127.0.0.0/8 (allow) [] to list loopback.auto > 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:987 Created ip list > localnet.auto default (deny) > 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:990 Adding > 10.1.1.4/255.255.255.0 (allow) to list localnet.auto > 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding > 192.168.42.0/24 (deny) [] to list lan > 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:1086 Adding > 192.168.42.0/24 (deny) to list lan > 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding > 192.168.42.42/32 (allow) [] to list lan > 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:1086 Adding > 192.168.42.42/32 (allow) to list lan > 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding > 192.0.2.0/24 (allow) [brian at templar.ath.cx] to list domains > > Where do all those IP addresses come from? Why are they there? > > > Ok, next thing I did was open up VS2008 and opened up > freeswitch-1.0.6/libs/esl/managed/managed_esl.2008.sln. No problems > there. I then hit compile and it worked without any errors. I had the > following uncommented: > > ThreadPool.QueueUserWorkItem(new WaitCallback(InboundMode)); > //ThreadPool.QueueUserWorkItem(new WaitCallback(OutboundModeSync)); > //ThreadPool.QueueUserWorkItem(new WaitCallback(OutboundModeAsync)); > > > > I then hit run and got the following error: > > An unhandled exception of type 'System.TypeInitializationException' > occurred in ManagedEsl.dll > Additional information: The type initializer for 'ESLPINVOKE' threw an > exception. > > > It breaks here: > > public ESLconnection(string host, string port, string password) : > this(ESLPINVOKE.new_ESLconnection__SWIG_1(host, port, password), > true) { > } > > > Output: > A first chance exception of type 'System.BadImageFormatException' > occurred in ManagedEsl.dll > A first chance exception of type 'System.TypeInitializationException' > occurred in ManagedEsl.dll > A first chance exception of type 'System.TypeInitializationException' > occurred in ManagedEsl.dll > An unhandled exception of type 'System.TypeInitializationException' > occurred in ManagedEsl.dll > > Additional information: The type initializer for 'ESLPINVOKE' threw an > exception. > > > Unfortunately I have no idea how to fix it! > > > Thanks, > > Oliver Schenk > > > > > > On 10/06/2010 8:48 PM, Oliver Schenk wrote: >> Hi, >> >> Thanks for everyone's reply. Great help. I have a basic IVR going >> when I >> dial a given extension. >> >> I have managed to clean out most of FreeSWITCH config and I've even >> written a little javascript file to which my call gets redirected >> to and >> it just says a small phrase using flite. I'm looking at buying more >> professional TTS voices such as Cebstral or Ivona. I'm hosting my >> freeswitch server in a Virtual Box instance running on Ubuntu Server >> 10.04. I do my testing from X-Lite running on the host computer. >> >> Anyway, that said, I think we will have to stick with Windows >> because: >> 1) Our existing railway SCADA server will be windows based. >> 2) The SCADA libraries are .NET based. >> 3) I don't want to have separate hardware just to run linux. >> 4) Our company isn't too good at supporting linux. >> >> >> My problem now is to try to somehow trigger freeswitch to make an >> outgoing call on demand... >> >> >> The flow of my logic is: >> >> 1) A critical alarm occurs on our SCADA system. >> 2) FreeSWITCH should be triggered to make an outgoing call. >> 3) Use a pre-recorded WAV or use TTS to tell the callee what the >> alarm is. >> >> >> The problem: >> >> I don't know how to "tell" freeswitch that it needs to make a call >> and >> maybe even pass some parameters. Given I have C# and SQL at my >> disposal, >> what should I use? A freeswitch javascript program that runs in an >> infinite loop and constantly scans for new alarms in a database? In >> the >> registry? In a folder using files? Using a http request? command >> line? >> >> In other words, how do I trigger a freeswitch javascript program >> externally? >> >> >> (By the way I tried to use the "managed" module that comes with >> freeswitch, but it just throws exceptions left right and center >> TypeInitializationException)... >> >> Anyone have some experience? >> >> >> Thanks, >> >> Oliver Schenk >> >> >> On 9/06/2010 10:43 PM, Tim St. Pierre wrote: >> >>> Hi Oliver, >>> >>> FreeSwitch will do whatever you tell it to do and no more. >>> >>> Here's a few suggestions though - >>> >>> Empty out the default dialplan directory. Don't throw those away, >>> as you may want to reference them >>> as examples, but move them somewhere else. >>> >>> Edit modules.conf.xml and comment or remove any modules that you >>> don't need. This will also save >>> memory and other resources. >>> >>> You can also disable all the SIP profiles except one. Pick one >>> that makes sense (either Internal or >>> External, it doesn't really matter that much), and edit it so that >>> it makes sense with respect to >>> your network. What is your topology? Will you just be setting >>> freeswitch up with a static IP >>> address and having calls sent to it by the main PBX? If that's >>> the case, you can disable a lot of >>> the STUN and uPNP functionality. Tell this profile to bind to the >>> IP and port that the PBX will >>> send the calls to. >>> >>> Then all you have to do, is create a very simple dialplan that >>> will answer an incoming call and >>> perform whatever task you want. You would essentially be starting >>> with a blank sheet, adding just >>> the functions that you want. >>> >>> Hope that makes sense. >>> >>> -Tim >>> >>> Oliver Schenk wrote: >>> >>> >>>> Hi All, >>>> >>>> The company I work for currently has quite an extensive phone >>>> network >>>> which gets carried between old analogue PABXes which also has an >>>> interface to IP based phones. All the phones in our office are >>>> connected >>>> via CAT5 cable using IP, however literally hundreds of phones out >>>> in the >>>> field (we operate railway infrastructure) are on standard voice >>>> analogue >>>> phones carried through fibreoptics. >>>> >>>> Anyway, I would like to use Freeswitch purely for its IVR and TTS >>>> abilities and nothing else. So basically I just need it to act >>>> like a >>>> slave to whatever IP phone network is already out there. I'm a bit >>>> worried if I fire up freeswitch it will hijack the phone network! >>>> >>>> All our phones are accessible via a 5 digit extension. I would like >>>> Freeswitch to be behind one of those ... say 12345. If anyone >>>> within our >>>> phone network dials 12345 then Freeswitch should answer. I guess my >>>> question is...how should I go about disabling most of FreeSwitch >>>> except >>>> it's ability to pick up the phone and speak IVR/TTS and make an >>>> outgoing >>>> call via the existing phone network? >>>> >>>> Any general pointers appreciated. >>>> >>>> >>>> Thanks, >>>> >>>> Oliver Schenk >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100610/51f732a0/attachment-0001.html From oliver.schenk at iinet.net.au Thu Jun 10 06:42:15 2010 From: oliver.schenk at iinet.net.au (Oliver Schenk) Date: Thu, 10 Jun 2010 21:42:15 +0800 Subject: [Freeswitch-users] Freeswitch in an existing phone network In-Reply-To: <4C10E667.8030001@iinet.net.au> References: <4C0E3FCE.1040803@iinet.net.au> <4C0FA8AD.8020907@communicatefreely.net> <4C10DF1F.5020802@iinet.net.au> <4C10E667.8030001@iinet.net.au> Message-ID: <4C10EBB7.7000501@iinet.net.au> Bastards lol I figured it out! In your C# projects it will only work if the Platform Target is set to x86. e.g. Go to project properties > Build tab > Platform Target: x86 You need to set this for ManagedEsl and ManagedEslTest if you are trying out the test program! Regards, Oliver On 10/06/2010 9:19 PM, Oliver Schenk wrote: > Just some more info regarding managed C# and ESL. This is what I did: > > Put the following line in modules.conf.xml: > > > > I also checked that the module is in the mod folder. > > > > I then started Freeswitch from cmd in windows using "Freeswitch.exe -nonat". > > There are no errors while it boots. I do however get all sorts of > strange IP addresses. My own LAN is a 10.1.1.* network, but I also get > IP addresses like: > > 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding > 172.16.0.0/12 (deny) [] to list wan.auto > 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding > 192.168.0.0/16 (deny) [] to list wan.auto > 2010-06-10 21:00:05.530374 [NOTICE] switch_core.c:970 Created ip list > nat.auto default (deny) > 2010-06-10 21:00:05.530374 [NOTICE] switch_core.c:972 Adding > 10.1.1.4/255.255.255.0 (deny) to list nat.auto > 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding 10.0.0.0/8 > (allow) [] to list nat.auto > 2010-06-10 21:00:05.530374 [NOTICE] switch_utils.c:195 Adding > 172.16.0.0/12 (allow) [] to list nat.auto > 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding > 192.168.0.0/16 (allow) [] to list nat.auto > 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:981 Created ip list > loopback.auto default (deny) > 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding > 127.0.0.0/8 (allow) [] to list loopback.auto > 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:987 Created ip list > localnet.auto default (deny) > 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:990 Adding > 10.1.1.4/255.255.255.0 (allow) to list localnet.auto > 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding > 192.168.42.0/24 (deny) [] to list lan > 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:1086 Adding > 192.168.42.0/24 (deny) to list lan > 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding > 192.168.42.42/32 (allow) [] to list lan > 2010-06-10 21:00:05.531374 [NOTICE] switch_core.c:1086 Adding > 192.168.42.42/32 (allow) to list lan > 2010-06-10 21:00:05.531374 [NOTICE] switch_utils.c:195 Adding > 192.0.2.0/24 (allow) [brian at templar.ath.cx] to list domains > > Where do all those IP addresses come from? Why are they there? > > > Ok, next thing I did was open up VS2008 and opened up > freeswitch-1.0.6/libs/esl/managed/managed_esl.2008.sln. No problems > there. I then hit compile and it worked without any errors. I had the > following uncommented: > > ThreadPool.QueueUserWorkItem(new WaitCallback(InboundMode)); > //ThreadPool.QueueUserWorkItem(new WaitCallback(OutboundModeSync)); > //ThreadPool.QueueUserWorkItem(new WaitCallback(OutboundModeAsync)); > > > > I then hit run and got the following error: > > An unhandled exception of type 'System.TypeInitializationException' > occurred in ManagedEsl.dll > Additional information: The type initializer for 'ESLPINVOKE' threw an > exception. > > > It breaks here: > > public ESLconnection(string host, string port, string password) : > this(ESLPINVOKE.new_ESLconnection__SWIG_1(host, port, password), true) { > } > > > Output: > A first chance exception of type 'System.BadImageFormatException' > occurred in ManagedEsl.dll > A first chance exception of type 'System.TypeInitializationException' > occurred in ManagedEsl.dll > A first chance exception of type 'System.TypeInitializationException' > occurred in ManagedEsl.dll > An unhandled exception of type 'System.TypeInitializationException' > occurred in ManagedEsl.dll > > Additional information: The type initializer for 'ESLPINVOKE' threw an > exception. > > > Unfortunately I have no idea how to fix it! > > > Thanks, > > Oliver Schenk > > > > > > On 10/06/2010 8:48 PM, Oliver Schenk wrote: > >> Hi, >> >> Thanks for everyone's reply. Great help. I have a basic IVR going when I >> dial a given extension. >> >> I have managed to clean out most of FreeSWITCH config and I've even >> written a little javascript file to which my call gets redirected to and >> it just says a small phrase using flite. I'm looking at buying more >> professional TTS voices such as Cebstral or Ivona. I'm hosting my >> freeswitch server in a Virtual Box instance running on Ubuntu Server >> 10.04. I do my testing from X-Lite running on the host computer. >> >> Anyway, that said, I think we will have to stick with Windows because: >> 1) Our existing railway SCADA server will be windows based. >> 2) The SCADA libraries are .NET based. >> 3) I don't want to have separate hardware just to run linux. >> 4) Our company isn't too good at supporting linux. >> >> >> My problem now is to try to somehow trigger freeswitch to make an >> outgoing call on demand... >> >> >> The flow of my logic is: >> >> 1) A critical alarm occurs on our SCADA system. >> 2) FreeSWITCH should be triggered to make an outgoing call. >> 3) Use a pre-recorded WAV or use TTS to tell the callee what the alarm is. >> >> >> The problem: >> >> I don't know how to "tell" freeswitch that it needs to make a call and >> maybe even pass some parameters. Given I have C# and SQL at my disposal, >> what should I use? A freeswitch javascript program that runs in an >> infinite loop and constantly scans for new alarms in a database? In the >> registry? In a folder using files? Using a http request? command line? >> >> In other words, how do I trigger a freeswitch javascript program externally? >> >> >> (By the way I tried to use the "managed" module that comes with >> freeswitch, but it just throws exceptions left right and center >> TypeInitializationException)... >> >> Anyone have some experience? >> >> >> Thanks, >> >> Oliver Schenk >> >> >> On 9/06/2010 10:43 PM, Tim St. Pierre wrote: >> >> >>> Hi Oliver, >>> >>> FreeSwitch will do whatever you tell it to do and no more. >>> >>> Here's a few suggestions though - >>> >>> Empty out the default dialplan directory. Don't throw those away, as you may want to reference them >>> as examples, but move them somewhere else. >>> >>> Edit modules.conf.xml and comment or remove any modules that you don't need. This will also save >>> memory and other resources. >>> >>> You can also disable all the SIP profiles except one. Pick one that makes sense (either Internal or >>> External, it doesn't really matter that much), and edit it so that it makes sense with respect to >>> your network. What is your topology? Will you just be setting freeswitch up with a static IP >>> address and having calls sent to it by the main PBX? If that's the case, you can disable a lot of >>> the STUN and uPNP functionality. Tell this profile to bind to the IP and port that the PBX will >>> send the calls to. >>> >>> Then all you have to do, is create a very simple dialplan that will answer an incoming call and >>> perform whatever task you want. You would essentially be starting with a blank sheet, adding just >>> the functions that you want. >>> >>> Hope that makes sense. >>> >>> -Tim >>> >>> Oliver Schenk wrote: >>> >>> >>> >>>> Hi All, >>>> >>>> The company I work for currently has quite an extensive phone network >>>> which gets carried between old analogue PABXes which also has an >>>> interface to IP based phones. All the phones in our office are connected >>>> via CAT5 cable using IP, however literally hundreds of phones out in the >>>> field (we operate railway infrastructure) are on standard voice analogue >>>> phones carried through fibreoptics. >>>> >>>> Anyway, I would like to use Freeswitch purely for its IVR and TTS >>>> abilities and nothing else. So basically I just need it to act like a >>>> slave to whatever IP phone network is already out there. I'm a bit >>>> worried if I fire up freeswitch it will hijack the phone network! >>>> >>>> All our phones are accessible via a 5 digit extension. I would like >>>> Freeswitch to be behind one of those ... say 12345. If anyone within our >>>> phone network dials 12345 then Freeswitch should answer. I guess my >>>> question is...how should I go about disabling most of FreeSwitch except >>>> it's ability to pick up the phone and speak IVR/TTS and make an outgoing >>>> call via the existing phone network? >>>> >>>> Any general pointers appreciated. >>>> >>>> >>>> Thanks, >>>> >>>> Oliver Schenk >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > From vetali100 at gmail.com Thu Jun 10 07:18:22 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Thu, 10 Jun 2010 17:18:22 +0300 Subject: [Freeswitch-users] LUA Callback - Help In-Reply-To: References: <20100610130042.4802f558@shadow.elt> Message-ID: Try to add short delay - session:sleep: ... session:setAutoHangup(false); session:sleep(1000); ... Sometimes I was getting session is not ready, and fixed this by adding short delay. Maybe you have same problem here. Regards, Vitalie 10 ???? 2010 ?. 16:20 ???????????? Aloysius Lloyd < lloyd.aloysius at sunteltech.ca> ???????: > VItalli, > > I try your suggestion. Now the script look like this > > > os.execute("sleep " .. tonumber(10)); > > pin_number_test="0511"; > out_gateway1="sofia/gateway/voipms/1"; > message_dir="/usr/local/freeswitch/sounds/en/us/callie/custom/callback/"; > > number_to_call= argv[1]; > > session = > freeswitch.Session("{ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); > session:setAutoHangup(false); > > if ( session:ready() ) then > session:sleep(1000); > if (session:answered()) then > session:sleep(1000); > session:streamFile(message_dir.."enter-dest.wav"); > end > end > > === > > Also I try 5 test . This time out of 5 test, wave file play only one time. > Other for 4 tests silent. > > Thanks > Lloyd > > > > 2010/6/10 Vitalii Colosov > > It looks OK what you wrote - we have almost same lua code. >> >> The only thing that I have in addition in my script is "if >> (session:answered())" construction. >> >> Try to write it like this: >> ... >> >> if ( session:ready() ) then >> session:sleep(1000); >> if (session:answered()) then >> session:sleep(1000); >> session:streamFile(message_dir.."enter-dest.wav"); >> end >> >> end >> ... >> >> Not sure why this should work and that does not, but this is how it works >> for me - worth trying at lest. >> >> >> Regards, >> Vitalie >> >> >> 10 ???? 2010 ?. 9:07 ???????????? Aloysius Lloyd < >> lloyd.aloysius at sunteltech.ca> ???????: >> >> >>> without bypass_media_after_bridge=true same results. >>> >>> Lloyd >>> >>> >>> >>> 2010/6/10 Aloysius Lloyd >>> >>> I try the bypass_media_after_bridge=true .. same results. >>>> >>>> Thanks >>>> Lloyd >>>> >>>> >>>> >>>> >>>> On Thu, Jun 10, 2010 at 2:00 AM, Sergey Scheglov wrote: >>>> >>>>> ? Thu, 10 Jun 2010 01:26:51 -0400 >>>>> Aloysius Lloyd wrote: >>>>> >>>>> >>>>> > session = >>>>> > >>>>> freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); >>>>> > session:setAutoHangup(false); >>>>> > >>>>> > Thanks >>>>> > Lloyd >>>>> >>>>> Try without bypass_media_after_bridge=true >>>>> >>>>> Regard >>>>> Sergey Scheglov >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100610/5b5bc1ba/attachment.html From yehavi.bourvine at gmail.com Thu Jun 10 07:32:40 2010 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 10 Jun 2010 17:32:40 +0300 Subject: [Freeswitch-users] Problem witg message-waiting NOTIFY for Audiocodes ATA Message-ID: Hello, Recently I've noticed that phones on our Audiocodes MP-124 adapter do not get the message waiting indicator. It seems like this behaviour has started a while ago after some upgrade (don't know whether it was Freeswitch or Audiocodes). The current versions are FS 1.0.6 and AudioCodes 5.6 >From various tests I've done with SIPSACK and tcpdumps I find the following: - after a Register command a "correct" notify is sent (i.e., the AudioCodes understands it). - later on "incorrect" notify's are sent which are rejected by the AC with "user not found". The "correct" notify is: NOTIFY sip:80633 at 10.64.0.8:5060 SIP/2.0 Via: SIP/2.0/UDP 10.64.0.3;rport;branch=z9hG4bKap2H5DSe6Zm7a Max-Forwards: 70 From: ;tag=vtgH30805ZH1Q To: ;tag=1c1710947583 Call-ID: 17109472461062010161640 at 10.64.0.8 CSeq: 131971366 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported 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, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=7200 Content-Type: application/simple-message-summary Content-Length: 106 Messages-Waiting: no Message-Account: sip:80633 at pbx-gr.XXX Voice-Message: 0/0 (0/0) While the "incorrect one" differs in the following two lines: From: >;tag=vtgH30805ZH1Q To: >;tag=1c1710947583 Taking the second one, changing the domain name and sending it with SIPSACK made it work correctly. (10.64.0.3 is the adderss of pbx-gr.XXX and 10.64.0.8 is the Audiocodes). Whose side is not working properly and should be asked to be fixed? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/8a570cc9/attachment-0001.html From lloyd.aloysius at sunteltech.ca Thu Jun 10 07:51:39 2010 From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd) Date: Thu, 10 Jun 2010 10:51:39 -0400 Subject: [Freeswitch-users] LUA Callback - Help In-Reply-To: References: <20100610130042.4802f558@shadow.elt> Message-ID: Thanks Vitalli. Increase the short delay solve the problem. I increase from 1000 to 2000. But when I answer the phone there is a silent before play the file because of the dealy. How to improve this? Is there any setting to make the session fast. Thanks Lloyd 2010/6/10 Vitalii Colosov > Try to add short delay - session:sleep: > > ... > session:setAutoHangup(false); > > session:sleep(1000); > ... > > Sometimes I was getting session is not ready, and fixed this by adding > short delay. > > Maybe you have same problem here. > > Regards, > Vitalie > > > > 10 ???? 2010 ?. 16:20 ???????????? Aloysius Lloyd < > lloyd.aloysius at sunteltech.ca> ???????: > > VItalli, >> >> I try your suggestion. Now the script look like this >> >> >> os.execute("sleep " .. tonumber(10)); >> >> pin_number_test="0511"; >> out_gateway1="sofia/gateway/voipms/1"; >> message_dir="/usr/local/freeswitch/sounds/en/us/callie/custom/callback/"; >> >> number_to_call= argv[1]; >> >> session = >> freeswitch.Session("{ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); >> session:setAutoHangup(false); >> >> if ( session:ready() ) then >> session:sleep(1000); >> if (session:answered()) then >> session:sleep(1000); >> session:streamFile(message_dir.."enter-dest.wav"); >> end >> end >> >> === >> >> Also I try 5 test . This time out of 5 test, wave file play only one >> time. Other for 4 tests silent. >> >> Thanks >> Lloyd >> >> >> >> 2010/6/10 Vitalii Colosov >> >> It looks OK what you wrote - we have almost same lua code. >>> >>> The only thing that I have in addition in my script is "if >>> (session:answered())" construction. >>> >>> Try to write it like this: >>> ... >>> >>> if ( session:ready() ) then >>> session:sleep(1000); >>> if (session:answered()) then >>> session:sleep(1000); >>> session:streamFile(message_dir.."enter-dest.wav"); >>> end >>> >>> end >>> ... >>> >>> Not sure why this should work and that does not, but this is how it works >>> for me - worth trying at lest. >>> >>> >>> Regards, >>> Vitalie >>> >>> >>> 10 ???? 2010 ?. 9:07 ???????????? Aloysius Lloyd < >>> lloyd.aloysius at sunteltech.ca> ???????: >>> >>> >>>> without bypass_media_after_bridge=true same results. >>>> >>>> Lloyd >>>> >>>> >>>> >>>> 2010/6/10 Aloysius Lloyd >>>> >>>> I try the bypass_media_after_bridge=true .. same results. >>>>> >>>>> Thanks >>>>> Lloyd >>>>> >>>>> >>>>> >>>>> >>>>> On Thu, Jun 10, 2010 at 2:00 AM, Sergey Scheglov wrote: >>>>> >>>>>> ? Thu, 10 Jun 2010 01:26:51 -0400 >>>>>> Aloysius Lloyd wrote: >>>>>> >>>>>> >>>>>> > session = >>>>>> > >>>>>> freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); >>>>>> > session:setAutoHangup(false); >>>>>> > >>>>>> > Thanks >>>>>> > Lloyd >>>>>> >>>>>> Try without bypass_media_after_bridge=true >>>>>> >>>>>> Regard >>>>>> Sergey Scheglov >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.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 >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/a0d9714d/attachment.html From lloyd.aloysius at gmail.com Thu Jun 10 07:59:08 2010 From: lloyd.aloysius at gmail.com (Aloysius Lloyd) Date: Thu, 10 Jun 2010 10:59:08 -0400 Subject: [Freeswitch-users] LUA - session:waitForAnswer - Error Message-ID: Hi All, In javascript the session.waitForAnswer(10000); working without any problem. But in Lua session:waitForAnswer(10000); throwing the following error. *2010-06-10 10:54:21.029607 [ERR] mod_lua.cpp:182 Error in waitForAnswer (arg 2), expected 'CoreSession *' got 'number' stack traceback:* What is the correct argument for this function in LUA. In wiki this function description is not available. Any help appreciate. Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/cf6fd0f2/attachment.html From vetali100 at gmail.com Thu Jun 10 08:51:05 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Thu, 10 Jun 2010 18:51:05 +0300 Subject: [Freeswitch-users] LUA Callback - Help In-Reply-To: References: <20100610130042.4802f558@shadow.elt> Message-ID: You can play with the delay values. Try this code: --------------------------------------------------- os.execute("sleep " .. tonumber(10)); pin_number_test="0511"; out_gateway1="sofia/gateway/voipms/1"; message_dir="/usr/local/freeswitch/sounds/en/us/callie/custom/callback/"; number_to_call= argv[1]; session = freeswitch.Session("{ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); session:setAutoHangup(false); session:sleep(400); if ( session:ready() ) then session:sleep(200); if (session:answered()) then session:sleep(200); session:streamFile(message_dir.."enter-dest.wav"); end end --------------------------------------------------- Tell me is this works. If not - try to modify delays (400,200,200), use different values less then 1 second. Regards, Vitalie 10 ???? 2010 ?. 17:51 ???????????? Aloysius Lloyd < lloyd.aloysius at sunteltech.ca> ???????: > Thanks Vitalli. > > Increase the short delay solve the problem. I increase from 1000 to 2000. > > But when I answer the phone there is a silent before play the file because > of the dealy. > > How to improve this? Is there any setting to make the session fast. > > > Thanks > Lloyd > > > 2010/6/10 Vitalii Colosov > >> Try to add short delay - session:sleep: >> >> ... >> session:setAutoHangup(false); >> >> session:sleep(1000); >> ... >> >> Sometimes I was getting session is not ready, and fixed this by adding >> short delay. >> >> Maybe you have same problem here. >> >> Regards, >> Vitalie >> >> >> >> 10 ???? 2010 ?. 16:20 ???????????? Aloysius Lloyd < >> lloyd.aloysius at sunteltech.ca> ???????: >> >> VItalli, >>> >>> I try your suggestion. Now the script look like this >>> >>> >>> os.execute("sleep " .. tonumber(10)); >>> >>> pin_number_test="0511"; >>> out_gateway1="sofia/gateway/voipms/1"; >>> message_dir="/usr/local/freeswitch/sounds/en/us/callie/custom/callback/"; >>> >>> number_to_call= argv[1]; >>> >>> session = >>> freeswitch.Session("{ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); >>> session:setAutoHangup(false); >>> >>> if ( session:ready() ) then >>> session:sleep(1000); >>> if (session:answered()) then >>> session:sleep(1000); >>> session:streamFile(message_dir.."enter-dest.wav"); >>> end >>> end >>> >>> === >>> >>> Also I try 5 test . This time out of 5 test, wave file play only one >>> time. Other for 4 tests silent. >>> >>> Thanks >>> Lloyd >>> >>> >>> >>> 2010/6/10 Vitalii Colosov >>> >>> It looks OK what you wrote - we have almost same lua code. >>>> >>>> The only thing that I have in addition in my script is "if >>>> (session:answered())" construction. >>>> >>>> Try to write it like this: >>>> ... >>>> >>>> if ( session:ready() ) then >>>> session:sleep(1000); >>>> if (session:answered()) then >>>> session:sleep(1000); >>>> session:streamFile(message_dir.."enter-dest.wav"); >>>> end >>>> >>>> end >>>> ... >>>> >>>> Not sure why this should work and that does not, but this is how it >>>> works for me - worth trying at lest. >>>> >>>> >>>> Regards, >>>> Vitalie >>>> >>>> >>>> 10 ???? 2010 ?. 9:07 ???????????? Aloysius Lloyd < >>>> lloyd.aloysius at sunteltech.ca> ???????: >>>> >>>> >>>>> without bypass_media_after_bridge=true same results. >>>>> >>>>> Lloyd >>>>> >>>>> >>>>> >>>>> 2010/6/10 Aloysius Lloyd >>>>> >>>>> I try the bypass_media_after_bridge=true .. same results. >>>>>> >>>>>> Thanks >>>>>> Lloyd >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Thu, Jun 10, 2010 at 2:00 AM, Sergey Scheglov wrote: >>>>>> >>>>>>> ? Thu, 10 Jun 2010 01:26:51 -0400 >>>>>>> Aloysius Lloyd wrote: >>>>>>> >>>>>>> >>>>>>> > session = >>>>>>> > >>>>>>> freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); >>>>>>> > session:setAutoHangup(false); >>>>>>> > >>>>>>> > Thanks >>>>>>> > Lloyd >>>>>>> >>>>>>> Try without bypass_media_after_bridge=true >>>>>>> >>>>>>> Regard >>>>>>> Sergey Scheglov >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.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 >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100610/93dc75bd/attachment-0001.html From infos at madovsky.org Thu Jun 10 08:59:00 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 11:59:00 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: <2FEECABF1A84404E8C751103FA4152CA@MOBILEE1705> Ok Rupa, when I tried the custom SQL example with gist option first It didn't recognize the @>'text'. I saw that the fields prefix and digits were not prefix_range type. I'm trying to reinstall the table again and test Regards F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 8:38 AM Subject: Re: [Freeswitch-users] mod_lcr Read the part about how to define the table + the gist index. The whole custom_sql thing assumes a familiarity with sql. you can choose to not have a digits_prefix column and just change the datatype of prefix to prefix. You can do what I did which is to have prefix be text and digits_prefix be of type prefix and a trigger to keep the two in sync. the key is that you are searching against the prefix column for which there is a GIST index. On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: ok thanks I will read again F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 3:09 AM Subject: Re: [Freeswitch-users] mod_lcr it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/166e6ff5/attachment.html From anthony.minessale at gmail.com Thu Jun 10 09:09:27 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Jun 2010 11:09:27 -0500 Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1276162496035-5162317.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> <1275859376110-5146705.post@n2.nabble.com> <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> <1275862175540-5146803.post@n2.nabble.com> <1276001266045-5153498.post@n2.nabble.com> <1276162496035-5162317.post@n2.nabble.com> Message-ID: The funny part is you are willing to go to all this effort to try to make us correct it because you know the other people wont. This type of finger pointing to RFC is typical to providers so now I have to do free work to investigate it for you. And I bet you are paying them money and not us, oh well, typical.....Don't worry I won't hold it against you. The philosophical question here is weather or not to send a requires timer in the reinvite, so ok. I can try to look it up for you but from a logical point of view, once you establish in the original invite that you are going to use session timers, every reinvite constitutes a timer refresh, so once you have agreed in the original invite to use timers it would stand to reason to require the timer option in every new re-invite since it represents a timer refresh. But, let's face it... Who cares right? its a meaningless param in an REINVITE. Most sip implementation wing it on most of these packet parsers. The sofia developers are very keen to actually follow all these silly RFC you are trying to quote and they have been relaxing it nonstop ever since because nobody really follows any rules in SIP they just use them to avoid having to patch their code. Especially in a deployed appliance in a telco. They are really telling you they can't support it because the code in their sip gateway fails when it sees this and they are too lazy to change it so they pray that some RFC will let them off the hook. Trust me, we've had worse. One place refused our calls because they did not like the syntax of our user agent string...... So bottom line is I would trust the default behavior of sofia-sip over most things any day but if they are really doing something wrong the problem would be in the sofia-sip library not in FreeSWITCH so I am not sure where to tell you to look in the code to correct it because we use that code asis and avoid hacking into it a all costs. Try using grep into the sofia-sip lib for the strings you know are meaningful. P.S All the event related things are fixed now, there was a small speed bump along the way this week. On Thu, Jun 10, 2010 at 4:34 AM, peely wrote: > > Hi, > > It's not a hijacked thread, I created this with an issue around uuid_media, > initially it was hanging, then in the latest git it receives a 4xx from the > far end due to the Requires: timer support bot being supported in the > reinvite transaction. I did anecdotally mention problems with events, which > I subsequently pulled off to another thread. > > I was looking at RFC 4028, in the examples it shows the refresh event not > providing the Required: timer header even for the refresh of the original > invite transaction, just the Supported: timer. There's nothing in RFC 3725 > regarding sesison timers in 3PCC but as the reinvite for the media > adjustment is a new transaction, I don't think it needs the Requires: timer > option to persist the timer support on the original dialogue. This is > certainly the behaviour of the previous equipment I had on this > interconnect > and it operates using session timers and 3PCC in several other environments > too. > > I'm not brilliant with C, but if you could please point me to the right > approach to removing the Requires: timer field in the header I'd be happy > to > test it and let you know if it causes any problems with rienvites across > various carriers and other platforms. > > > > Cheers, > > > Neil. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5162317.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100610/98197aa7/attachment-0001.html From anthony.minessale at gmail.com Thu Jun 10 09:17:10 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Jun 2010 11:17:10 -0500 Subject: [Freeswitch-users] LUA - session:waitForAnswer - Error In-Reply-To: References: Message-ID: it as no time arguments in the lua version only an optional alternate channel like session2:waitForAnswer(session) On Thu, Jun 10, 2010 at 9:59 AM, Aloysius Lloyd wrote: > Hi All, > > In javascript the session.waitForAnswer(10000); working without any > problem. > > But in Lua session:waitForAnswer(10000); throwing the following error. > > > *2010-06-10 10:54:21.029607 [ERR] mod_lua.cpp:182 Error in waitForAnswer > (arg 2), expected 'CoreSession *' got 'number' > stack traceback:* > > > What is the correct argument for this function in LUA. In wiki this > function description is not available. > > Any help appreciate. > > Thanks > Lloyd > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100610/3913dce8/attachment.html From infos at madovsky.org Thu Jun 10 09:24:06 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 12:24:06 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: Rupa, I'd like to test PREFIX module form postgresql. so in your postgresql sql I can see : CREATE TABLE lcr ( ... digits NUMERIC(20, 0), .... prefix VARCHAR(16) NOT NULL DEFAULT '', ... ); but in your PREFIX postgresql module example it's written : create table rates (id serial not null, prefix prefix_range, [...]); create index rates_prefix_idx on rates using gist (prefix gist_prefix_range_ops); insert [...] select * from rates where prefix @> '16666666666'; and in the custom example :...WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits_prefix @> '%q' .... My question is how to uise the lcr table with PREFIX module of postgresql ? Thaks Franck ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 8:38 AM Subject: Re: [Freeswitch-users] mod_lcr Read the part about how to define the table + the gist index. The whole custom_sql thing assumes a familiarity with sql. you can choose to not have a digits_prefix column and just change the datatype of prefix to prefix. You can do what I did which is to have prefix be text and digits_prefix be of type prefix and a trigger to keep the two in sync. the key is that you are searching against the prefix column for which there is a GIST index. On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: ok thanks I will read again F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 3:09 AM Subject: Re: [Freeswitch-users] mod_lcr it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/7b9c7c86/attachment.html From anthony.minessale at gmail.com Thu Jun 10 09:24:31 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Jun 2010 11:24:31 -0500 Subject: [Freeswitch-users] JavaScript control Transfer In-Reply-To: <1276128865845-5161043.post@n2.nabble.com> References: <760778166.8.1276124703365.JavaMail.root@srvr12.remotelinkml.com> <1276128865845-5161043.post@n2.nabble.com> Message-ID: session.execute("transfer", ":'' inline"); On Wed, Jun 9, 2010 at 7:14 PM, David Swardstrom wrote: > > I really want to do the transfer and plan to exit() after the transfer > takes > place. > What I have is a string like the following: "conf-ivr.js 6". > The conf-ivr.js is my version of the conf-ivr.js example. > The value "6" indicates a sub-type of conference so that we can support > different flavors. > I guess I could use the following: > session.execute("transfer", "conf-ivr6"); > Then put the following into the xml: > > > > > > > What I hoped for was a way to avoid putting specific stuff for each > application into the XML. > I just wanted to be able to provide a new JS application or a new flavor of > a JS application > without having to also modify the XML and get the XML reloaded into > FreeSwitch. > > Is it possible to pass parameters via the transfer so the XML would look > like this: > > > > > > > There was a hint about using something called "inline" which indicated that > an XML dialplan > would not be needed. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/JavaScript-control-Transfer-tp5160890p5161043.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100610/3127ff14/attachment-0001.html From infos at madovsky.org Thu Jun 10 09:30:42 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 12:30:42 -0400 Subject: [Freeswitch-users] uuid_media hangs References: <1275837444425-5145657.post@n2.nabble.com><1275857911630-5146647.post@n2.nabble.com><1275859376110-5146705.post@n2.nabble.com><9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org><1275862175540-5146803.post@n2.nabble.com><1276001266045-5153498.post@n2.nabble.com><1276162496035-5162317.post@n2.nabble.com> Message-ID: <194C8CC139AE418D9A819801C2D7B5B3@MOBILEE1705> > are very keen to actually follow all these silly RFC I agree, usually RFC are created first by university so by students... ----- Original Message ----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 12:09 PM Subject: Re: [Freeswitch-users] uuid_media hangs The funny part is you are willing to go to all this effort to try to make us correct it because you know the other people wont. This type of finger pointing to RFC is typical to providers so now I have to do free work to investigate it for you. And I bet you are paying them money and not us, oh well, typical.....Don't worry I won't hold it against you. The philosophical question here is weather or not to send a requires timer in the reinvite, so ok. I can try to look it up for you but from a logical point of view, once you establish in the original invite that you are going to use session timers, every reinvite constitutes a timer refresh, so once you have agreed in the original invite to use timers it would stand to reason to require the timer option in every new re-invite since it represents a timer refresh. But, let's face it... Who cares right? its a meaningless param in an REINVITE. Most sip implementation wing it on most of these packet parsers. The sofia developers are very keen to actually follow all these silly RFC you are trying to quote and they have been relaxing it nonstop ever since because nobody really follows any rules in SIP they just use them to avoid having to patch their code. Especially in a deployed appliance in a telco. They are really telling you they can't support it because the code in their sip gateway fails when it sees this and they are too lazy to change it so they pray that some RFC will let them off the hook. Trust me, we've had worse. One place refused our calls because they did not like the syntax of our user agent string...... So bottom line is I would trust the default behavior of sofia-sip over most things any day but if they are really doing something wrong the problem would be in the sofia-sip library not in FreeSWITCH so I am not sure where to tell you to look in the code to correct it because we use that code asis and avoid hacking into it a all costs. Try using grep into the sofia-sip lib for the strings you know are meaningful. P.S All the event related things are fixed now, there was a small speed bump along the way this week. On Thu, Jun 10, 2010 at 4:34 AM, peely wrote: Hi, It's not a hijacked thread, I created this with an issue around uuid_media, initially it was hanging, then in the latest git it receives a 4xx from the far end due to the Requires: timer support bot being supported in the reinvite transaction. I did anecdotally mention problems with events, which I subsequently pulled off to another thread. I was looking at RFC 4028, in the examples it shows the refresh event not providing the Required: timer header even for the refresh of the original invite transaction, just the Supported: timer. There's nothing in RFC 3725 regarding sesison timers in 3PCC but as the reinvite for the media adjustment is a new transaction, I don't think it needs the Requires: timer option to persist the timer support on the original dialogue. This is certainly the behaviour of the previous equipment I had on this interconnect and it operates using session timers and 3PCC in several other environments too. I'm not brilliant with C, but if you could please point me to the right approach to removing the Requires: timer field in the header I'd be happy to test it and let you know if it causes any problems with rienvites across various carriers and other platforms. Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5162317.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/0f5b22ee/attachment.html From kris at kriskinc.com Thu Jun 10 10:00:39 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 10 Jun 2010 13:00:39 -0400 Subject: [Freeswitch-users] uuid_media hangs Message-ID: <0a50a72df10659f4cba4e2f7790e4a1a@mail.gmail.com> That's completely ridiculous. -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces at lists.freeswitch.org < freeswitch-users-bounces at lists.freeswitch.org> *To*: freeswitch-users at lists.freeswitch.org < freeswitch-users at lists.freeswitch.org> *Sent*: Thu Jun 10 12:30:42 2010 *Subject*: Re: [Freeswitch-users] uuid_media hangs > are very keen to actually follow all these silly RFC I agree, usually RFC are created first by university so by students... ----- Original Message ----- *From:* Anthony Minessale *To:* freeswitch-users at lists.freeswitch.org *Sent:* Thursday, June 10, 2010 12:09 PM *Subject:* Re: [Freeswitch-users] uuid_media hangs The funny part is you are willing to go to all this effort to try to make us correct it because you know the other people wont. This type of finger pointing to RFC is typical to providers so now I have to do free work to investigate it for you. And I bet you are paying them money and not us, oh well, typical.....Don't worry I won't hold it against you. The philosophical question here is weather or not to send a requires timer in the reinvite, so ok. I can try to look it up for you but from a logical point of view, once you establish in the original invite that you are going to use session timers, every reinvite constitutes a timer refresh, so once you have agreed in the original invite to use timers it would stand to reason to require the timer option in every new re-invite since it represents a timer refresh. But, let's face it... Who cares right? its a meaningless param in an REINVITE. Most sip implementation wing it on most of these packet parsers. The sofia developers are very keen to actually follow all these silly RFC you are trying to quote and they have been relaxing it nonstop ever since because nobody really follows any rules in SIP they just use them to avoid having to patch their code. Especially in a deployed appliance in a telco. They are really telling you they can't support it because the code in their sip gateway fails when it sees this and they are too lazy to change it so they pray that some RFC will let them off the hook. Trust me, we've had worse. One place refused our calls because they did not like the syntax of our user agent string...... So bottom line is I would trust the default behavior of sofia-sip over most things any day but if they are really doing something wrong the problem would be in the sofia-sip library not in FreeSWITCH so I am not sure where to tell you to look in the code to correct it because we use that code asis and avoid hacking into it a all costs. Try using grep into the sofia-sip lib for the strings you know are meaningful. P.S All the event related things are fixed now, there was a small speed bump along the way this week. On Thu, Jun 10, 2010 at 4:34 AM, peely wrote: > > Hi, > > It's not a hijacked thread, I created this with an issue around uuid_media, > initially it was hanging, then in the latest git it receives a 4xx from the > far end due to the Requires: timer support bot being supported in the > reinvite transaction. I did anecdotally mention problems with events, which > I subsequently pulled off to another thread. > > I was looking at RFC 4028, in the examples it shows the refresh event not > providing the Required: timer header even for the refresh of the original > invite transaction, just the Supported: timer. There's nothing in RFC 3725 > regarding sesison timers in 3PCC but as the reinvite for the media > adjustment is a new transaction, I don't think it needs the Requires: timer > option to persist the timer support on the original dialogue. This is > certainly the behaviour of the previous equipment I had on this > interconnect > and it operates using session timers and 3PCC in several other environments > too. > > I'm not brilliant with C, but if you could please point me to the right > approach to removing the Requires: timer field in the header I'd be happy > to > test it and let you know if it causes any problems with rienvites across > various carriers and other platforms. > > > > Cheers, > > > Neil. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5162317.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/b62b3bdb/attachment-0001.html From lloyd.aloysius at sunteltech.ca Thu Jun 10 10:00:35 2010 From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd) Date: Thu, 10 Jun 2010 13:00:35 -0400 Subject: [Freeswitch-users] LUA - session:waitForAnswer - Error In-Reply-To: References: Message-ID: Thanks Anthony. Lloyd On Thu, Jun 10, 2010 at 12:17 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > it as no time arguments in the lua version only an optional alternate > channel > > like session2:waitForAnswer(session) > > On Thu, Jun 10, 2010 at 9:59 AM, Aloysius Lloyd wrote: > >> Hi All, >> >> In javascript the session.waitForAnswer(10000); working without any >> problem. >> >> But in Lua session:waitForAnswer(10000); throwing the following error. >> >> >> *2010-06-10 10:54:21.029607 [ERR] mod_lua.cpp:182 Error in waitForAnswer >> (arg 2), expected 'CoreSession *' got 'number' >> stack traceback:* >> >> >> What is the correct argument for this function in LUA. In wiki this >> function description is not available. >> >> Any help appreciate. >> >> Thanks >> Lloyd >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/6cd7ec79/attachment.html From neilp at cs.stanford.edu Thu Jun 10 10:01:29 2010 From: neilp at cs.stanford.edu (Neil Patel) Date: Thu, 10 Jun 2010 10:01:29 -0700 Subject: [Freeswitch-users] Lua script hangup detect In-Reply-To: <5d2828f1002111351s5f0cdff2odb6b35fa9be9eb32@mail.gmail.com> References: <314dc3f81002100117h6b1bc49cna989624a1ae4e0e2@mail.gmail.com> <314dc3f81002100800o41fd6561w394d5e83e69d3d80@mail.gmail.com> <4B72DFF9.4040402@gmail.com> <1b46b4e81002100932x7af486ekd4b05f4750cace20@mail.gmail.com> <4B72F2A8.4070503@gmail.com> <4CA5432B-5F6E-438E-B6EF-3277847E7922@jerris.com> <4B73B57E.9040901@gmail.com> <4B73C640.20700@gmail.com> <5d2828f1002111351s5f0cdff2odb6b35fa9be9eb32@mail.gmail.com> Message-ID: I've noticed that session:ready() stays set to true when initiating a call from lua and the end point never picks up or rejects the call: session = freeswitch.Session(DIALSTRING) session:setVariable("caller_id_number", phonenum) session:setVariable("playback_terminators", "#"); session:setHangupHook("hangup"); while (session:ready() == true) do ... How do I detect if the call was never accepted? -Neil On Thu, Feb 11, 2010 at 2:51 PM, Mike van Lammeren wrote: > D'oh! > > I am currently working on a project that uses Lua and the native MySQL > driver, so as soon as I read this comment, I decided that I had better do a > bit of research. > > I wrote a Lua test script that makes 10 queries against a MySQL database, > then ran it repeatedly. > > My results show that "leaks like a sieve" is quite correct. To me, it > doesn't look like any memory is released, ever. After only 5000 queries or > so, the memory allocated for FreeSWITCH balloons from 15 Mb to 50 Mb, and > never goes back down. > > Thanks, Brian, for the heads up! > > Mike van Lammeren > > > On Thu, Feb 11, 2010 at 8:26 AM, Brian West wrote: > >> Yes ODBC doesn't seem to leak... while the native one leaks like a sieve. >> >> /b >> >> On Feb 11, 2010, at 2:56 AM, Nazim Agabekov wrote: >> >> > I'm using luasql with ODBC MySQL driver in production. I've never tried >> > to use luasql with "native" mysql driver, but ODBC one works great. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100610/ac7772db/attachment.html From infos at madovsky.org Thu Jun 10 10:05:46 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 13:05:46 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: <8F25FD155CC14B619DD2873F58822BB3@MOBILEE1705> Sorry for my confusio Rupa, but when you say AND digits_prefix @> '%q' in your custom example is digits_prefix a field or other ? if it's a field that make the SQL request ambiguous since it doesn't contains any alias. Thanks F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 8:38 AM Subject: Re: [Freeswitch-users] mod_lcr Read the part about how to define the table + the gist index. The whole custom_sql thing assumes a familiarity with sql. you can choose to not have a digits_prefix column and just change the datatype of prefix to prefix. You can do what I did which is to have prefix be text and digits_prefix be of type prefix and a trigger to keep the two in sync. the key is that you are searching against the prefix column for which there is a GIST index. On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: ok thanks I will read again F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 3:09 AM Subject: Re: [Freeswitch-users] mod_lcr it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/3d12c839/attachment-0001.html From infos at madovsky.org Thu Jun 10 10:17:20 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 13:17:20 -0400 Subject: [Freeswitch-users] uuid_media hangs References: <0a50a72df10659f4cba4e2f7790e4a1a@mail.gmail.com> Message-ID: <99E77367AFAE41BF8764345DBFBA23FA@MOBILEE1705> why ? ----- Original Message ----- From: Kristian Kielhofner To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 1:00 PM Subject: Re: [Freeswitch-users] uuid_media hangs That's completely ridiculous. -- Kristian Kielhofner http://blog.krisk.org ------------------------------------------------------------------------------ From: freeswitch-users-bounces at lists.freeswitch.org To: freeswitch-users at lists.freeswitch.org Sent: Thu Jun 10 12:30:42 2010 Subject: Re: [Freeswitch-users] uuid_media hangs > are very keen to actually follow all these silly RFC I agree, usually RFC are created first by university so by students... ----- Original Message ----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 12:09 PM Subject: Re: [Freeswitch-users] uuid_media hangs The funny part is you are willing to go to all this effort to try to make us correct it because you know the other people wont. This type of finger pointing to RFC is typical to providers so now I have to do free work to investigate it for you. And I bet you are paying them money and not us, oh well, typical.....Don't worry I won't hold it against you. The philosophical question here is weather or not to send a requires timer in the reinvite, so ok. I can try to look it up for you but from a logical point of view, once you establish in the original invite that you are going to use session timers, every reinvite constitutes a timer refresh, so once you have agreed in the original invite to use timers it would stand to reason to require the timer option in every new re-invite since it represents a timer refresh. But, let's face it... Who cares right? its a meaningless param in an REINVITE. Most sip implementation wing it on most of these packet parsers. The sofia developers are very keen to actually follow all these silly RFC you are trying to quote and they have been relaxing it nonstop ever since because nobody really follows any rules in SIP they just use them to avoid having to patch their code. Especially in a deployed appliance in a telco. They are really telling you they can't support it because the code in their sip gateway fails when it sees this and they are too lazy to change it so they pray that some RFC will let them off the hook. Trust me, we've had worse. One place refused our calls because they did not like the syntax of our user agent string...... So bottom line is I would trust the default behavior of sofia-sip over most things any day but if they are really doing something wrong the problem would be in the sofia-sip library not in FreeSWITCH so I am not sure where to tell you to look in the code to correct it because we use that code asis and avoid hacking into it a all costs. Try using grep into the sofia-sip lib for the strings you know are meaningful. P.S All the event related things are fixed now, there was a small speed bump along the way this week. On Thu, Jun 10, 2010 at 4:34 AM, peely wrote: Hi, It's not a hijacked thread, I created this with an issue around uuid_media, initially it was hanging, then in the latest git it receives a 4xx from the far end due to the Requires: timer support bot being supported in the reinvite transaction. I did anecdotally mention problems with events, which I subsequently pulled off to another thread. I was looking at RFC 4028, in the examples it shows the refresh event not providing the Required: timer header even for the refresh of the original invite transaction, just the Supported: timer. There's nothing in RFC 3725 regarding sesison timers in 3PCC but as the reinvite for the media adjustment is a new transaction, I don't think it needs the Requires: timer option to persist the timer support on the original dialogue. This is certainly the behaviour of the previous equipment I had on this interconnect and it operates using session timers and 3PCC in several other environments too. I'm not brilliant with C, but if you could please point me to the right approach to removing the Requires: timer field in the header I'd be happy to test it and let you know if it causes any problems with rienvites across various carriers and other platforms. Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5162317.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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/20100610/d4080223/attachment.html From brian at freeswitch.org Thu Jun 10 10:26:30 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Jun 2010 12:26:30 -0500 Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <99E77367AFAE41BF8764345DBFBA23FA@MOBILEE1705> References: <0a50a72df10659f4cba4e2f7790e4a1a@mail.gmail.com> <99E77367AFAE41BF8764345DBFBA23FA@MOBILEE1705> Message-ID: Because RFC's are usually written by people that are very well versed in the field they are writing about... Granted I don't know exactly what they were smoking or drinking when they came up with SIP. /b On Jun 10, 2010, at 12:17 PM, Madovsky wrote: > why ? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/d5ef0457/attachment.html From msc at freeswitch.org Thu Jun 10 10:33:59 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Jun 2010 10:33:59 -0700 Subject: [Freeswitch-users] LUA Callback - Help In-Reply-To: References: <20100610130042.4802f558@shadow.elt> Message-ID: You do realize that you don't even need Lua to accomplish what you want to do, right? Instead of launching "luarun callback.lua ${caller_id_number}" as your API you can just do an originate: Create an extension 12345 (or pick a different number) and have the dialplan apps in that extension do all the stuff you are trying to do. Example: Remember the golden rule: Use the dialplan wherever possible because it will always be more efficient than a scripting language. (BTW, when the FS book comes out you'll see more about this topic.) HTH and happy dialing! -MC 2010/6/10 Aloysius Lloyd > Thanks Vitalli. > > Increase the short delay solve the problem. I increase from 1000 to 2000. > > But when I answer the phone there is a silent before play the file because > of the dealy. > > How to improve this? Is there any setting to make the session fast. > > > Thanks > Lloyd > > > 2010/6/10 Vitalii Colosov > >> Try to add short delay - session:sleep: >> >> ... >> session:setAutoHangup(false); >> >> session:sleep(1000); >> ... >> >> Sometimes I was getting session is not ready, and fixed this by adding >> short delay. >> >> Maybe you have same problem here. >> >> Regards, >> Vitalie >> >> >> >> 10 ???? 2010 ?. 16:20 ???????????? Aloysius Lloyd < >> lloyd.aloysius at sunteltech.ca> ???????: >> >> VItalli, >>> >>> I try your suggestion. Now the script look like this >>> >>> >>> os.execute("sleep " .. tonumber(10)); >>> >>> pin_number_test="0511"; >>> out_gateway1="sofia/gateway/voipms/1"; >>> message_dir="/usr/local/freeswitch/sounds/en/us/callie/custom/callback/"; >>> >>> number_to_call= argv[1]; >>> >>> session = >>> freeswitch.Session("{ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); >>> session:setAutoHangup(false); >>> >>> if ( session:ready() ) then >>> session:sleep(1000); >>> if (session:answered()) then >>> session:sleep(1000); >>> session:streamFile(message_dir.."enter-dest.wav"); >>> end >>> end >>> >>> === >>> >>> Also I try 5 test . This time out of 5 test, wave file play only one >>> time. Other for 4 tests silent. >>> >>> Thanks >>> Lloyd >>> >>> >>> >>> 2010/6/10 Vitalii Colosov >>> >>> It looks OK what you wrote - we have almost same lua code. >>>> >>>> The only thing that I have in addition in my script is "if >>>> (session:answered())" construction. >>>> >>>> Try to write it like this: >>>> ... >>>> >>>> if ( session:ready() ) then >>>> session:sleep(1000); >>>> if (session:answered()) then >>>> session:sleep(1000); >>>> session:streamFile(message_dir.."enter-dest.wav"); >>>> end >>>> >>>> end >>>> ... >>>> >>>> Not sure why this should work and that does not, but this is how it >>>> works for me - worth trying at lest. >>>> >>>> >>>> Regards, >>>> Vitalie >>>> >>>> >>>> 10 ???? 2010 ?. 9:07 ???????????? Aloysius Lloyd < >>>> lloyd.aloysius at sunteltech.ca> ???????: >>>> >>>> >>>>> without bypass_media_after_bridge=true same results. >>>>> >>>>> Lloyd >>>>> >>>>> >>>>> >>>>> 2010/6/10 Aloysius Lloyd >>>>> >>>>> I try the bypass_media_after_bridge=true .. same results. >>>>>> >>>>>> Thanks >>>>>> Lloyd >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Thu, Jun 10, 2010 at 2:00 AM, Sergey Scheglov wrote: >>>>>> >>>>>>> ? Thu, 10 Jun 2010 01:26:51 -0400 >>>>>>> Aloysius Lloyd wrote: >>>>>>> >>>>>>> >>>>>>> > session = >>>>>>> > >>>>>>> freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); >>>>>>> > session:setAutoHangup(false); >>>>>>> > >>>>>>> > Thanks >>>>>>> > Lloyd >>>>>>> >>>>>>> Try without bypass_media_after_bridge=true >>>>>>> >>>>>>> Regard >>>>>>> Sergey Scheglov >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.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 >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100610/c267e384/attachment-0001.html From infos at madovsky.org Thu Jun 10 10:44:34 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 13:44:34 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: Ok I understand now. but where to create digist_prefix in lcr table ? the end is ok ? Thanks F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 8:38 AM Subject: Re: [Freeswitch-users] mod_lcr Read the part about how to define the table + the gist index. The whole custom_sql thing assumes a familiarity with sql. you can choose to not have a digits_prefix column and just change the datatype of prefix to prefix. You can do what I did which is to have prefix be text and digits_prefix be of type prefix and a trigger to keep the two in sync. the key is that you are searching against the prefix column for which there is a GIST index. On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: ok thanks I will read again F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 3:09 AM Subject: Re: [Freeswitch-users] mod_lcr it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/d95fa5f1/attachment.html From infos at madovsky.org Thu Jun 10 10:45:27 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 13:45:27 -0400 Subject: [Freeswitch-users] uuid_media hangs References: <0a50a72df10659f4cba4e2f7790e4a1a@mail.gmail.com><99E77367AFAE41BF8764345DBFBA23FA@MOBILEE1705> Message-ID: <6E72182B398947D9B2FE185DB07D266B@MOBILEE1705> That makes sens.. .:D ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 1:26 PM Subject: Re: [Freeswitch-users] uuid_media hangs Because RFC's are usually written by people that are very well versed in the field they are writing about... Granted I don't know exactly what they were smoking or drinking when they came up with SIP. /b On Jun 10, 2010, at 12:17 PM, Madovsky wrote: why ? ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/a347a193/attachment.html From helmut.kuper at ewetel.de Thu Jun 10 10:50:13 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 10 Jun 2010 19:50:13 +0200 Subject: [Freeswitch-users] updated sqlite3? Message-ID: <4C1125D5.1070004@ewetel.de> Hello, today I played around with lua scripts using lua-sqlite3 module. I found that the delete statement of the sqlite3 version (3.1.3?) FS is using is not able to handle LIMIT in DELETE statements. Also the centos sqlite3 cli tool (v3.3.6) doesn't do. I found that sqlite3 version 3.6.23 does the trick as long as you use SQLITE_ENABLE_UPDATE_DELETE_LIMIT during compile time. So any chance to get FS sqlite sources updated? As long as not, can I riskless update FS's sqlite source to 3.6.23 and recompile FS? regards Helmut From freeswitch at peely.com Thu Jun 10 12:20:58 2010 From: freeswitch at peely.com (peely) Date: Thu, 10 Jun 2010 12:20:58 -0700 (PDT) Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> <1275859376110-5146705.post@n2.nabble.com> <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> <1275862175540-5146803.post@n2.nabble.com> <1276001266045-5153498.post@n2.nabble.com> <1276162496035-5162317.post@n2.nabble.com> Message-ID: <1276197658427-5164872.post@n2.nabble.com> Hi Anthony, You are right to a degree, being that the carrier in question is the national PTT, and their equipment is Acme, they are in the belief that their equipment is operating normally. Added to this the fact that we are the only interconnect customer who is actually wanting to do this (new tests had to be added to the sandbox testing to accommodate this) and are relatively small at this point then they have little interest in fixing the issue. The standards are vague at best, and little cross-referencing of existing standards happens when dreaming up a new one. You only have to look at the 3GPP specifications, which take a handful of mainly SIP RFCs and turn them into 000's of pages of standard to see how they need to appear to be specific. Having worked for a relatively small closed-source application server vendor previously, I know what it's like to be on the losing end of a standards argument when doing interop' against a larger vendor. Anyhow, thanks for the pointers and I'll take a look at Sofia and report something to them if need be. I think my first option may be to disable session timers on the test route initially, just to see if there's anything else afoot. Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5164872.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vetali100 at gmail.com Thu Jun 10 12:27:07 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Thu, 10 Jun 2010 22:27:07 +0300 Subject: [Freeswitch-users] Callback with bypass media Message-ID: Hi, Could you please advice if this callback scenario can be achieved? Freeswitch calls one number via external provider. After it answered, it gets the digits and dials another number via external provider. After second phone is answered, it bridge both sessions. Is it possible that from this moment media will flow directly from one sip phone to another, bypassing freeswitch? And if yes, which command (in Lua preferably)? I tried to use the following code after getting the digits from first phone: ... calling_session:setVariable("bypass_media", "true"); called_session = freeswitch.Session(some_called_string, calling_session); ... But as per log, the INVITE to the second phone contains FreeSWITCH's IP address in SDP section, and not first phone's IP address. Looks like bypass_media is not working as I would expect in this case. Thanks, Vitalie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/ddbccfe0/attachment.html From david.ponzone at gmail.com Thu Jun 10 13:10:03 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 10 Jun 2010 22:10:03 +0200 Subject: [Freeswitch-users] Callback with bypass media In-Reply-To: References: Message-ID: <5DFD32DE-05E2-4501-B646-A1BF7D101F69@gmail.com> Vitalii, You said you want to call both numbers via external provider. So you want the media to flow from external provider to external provider, not from SIP phone to SIP phone. This should work, but I must admit I never did that with 2 outbound legs. If it doesn't, I would recommend you pastebin the conf and the trace of a call attempt. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 10/06/2010 ? 21:27, Vitalii Colosov a ?crit : > Hi, > Could you please advice if this callback scenario can be achieved? > > Freeswitch calls one number via external provider. > After it answered, it gets the digits and dials another number > via external provider. > After second phone is answered, it bridge both sessions. > > Is it possible that from this moment media will flow directly > from one sip phone to another, bypassing freeswitch? > And if yes, which command (in Lua preferably)? > > > > I tried to use the following code after getting the digits from > first phone: > > ... > calling_session:setVariable("bypass_media", "true"); > called_session = freeswitch.Session(some_called_string, > calling_session); > ... > > > But as per log, the INVITE to the second phone contains FreeSWITCH's > IP address in SDP section, and not first phone's IP address. > Looks like bypass_media is not working as I would expect in this case. > > > > Thanks, > Vitalie > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/f574b183/attachment.html From rupa at rupa.com Thu Jun 10 13:10:41 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 10 Jun 2010 15:10:41 -0500 Subject: [Freeswitch-users] mod_lcr In-Reply-To: References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> <763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: What I use that differs from default: digits | text | not null digits_prefix | prefix_range | not null Indexes: "idx_prefix" gist (digits_prefix gist_prefix_range_ops) I keep digits around so I can test both default behavior and new behavior. You only really NEED the second one. On insert I set them to the same value (eg: 12145551212). End is fine. Doesn't really matter. On Thu, Jun 10, 2010 at 12:44 PM, Madovsky wrote: > Ok I understand now. > but where to create digist_prefix in lcr table ? the end is ok ? > > Thanks > > F > > ----- Original Message ----- > *From:* Rupa Schomaker > *To:* freeswitch-users > *Sent:* Thursday, June 10, 2010 8:38 AM > *Subject:* Re: [Freeswitch-users] mod_lcr > > Read the part about how to define the table + the gist index. The whole > custom_sql thing assumes a familiarity with sql. you can choose to not have > a digits_prefix column and just change the datatype of prefix to prefix. > You can do what I did which is to have prefix be text and digits_prefix be > of type prefix and a trigger to keep the two in sync. the key is that you > are searching against the prefix column for which there is a GIST index. > > On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: > >> ok thanks I will read again >> >> F >> >> ----- Original Message ----- >> *From:* Nandy Dagondon >> *To:* freeswitch-users at lists.freeswitch.org >> *Sent:* Thursday, June 10, 2010 3:09 AM >> *Subject:* Re: [Freeswitch-users] mod_lcr >> >> it's the digits_prefix in the WHERE clause that's causing the error. >> >> ur question re prefix+digits, it's explained in the Custom SQL portion in >> the wiki. >> >> -nandy >> >> >> On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: >> >>> ok so it needs also the alias l.digits in the condition I think. >>> I'm a little confused about digits and prefix. >>> if I check a number with the country code is it need to join >>> prefix+digits ? how with this kinkd of sql request ? >>> >>> Thanks >>> >>> F >>> >>> ----- Original Message ----- >>> *From:* Nandy Dagondon >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Sent:* Thursday, June 10, 2010 2:44 AM >>> *Subject:* Re: [Freeswitch-users] mod_lcr >>> >>> i think it's a typo. i changed digits_prefix to digits. to be sure, pls >>> check the CREATE TABLE entries. >>> -nandy >>> >>> >>> On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: >>> >>>> I'm experimenting with mod_lcr with postgresql (8.4.4) >>>> there s an example of custom sql on wiki below : >>>> >>>> >>>> >>>> >>>> >>>> >>>> however the query failed cause of digits_prefix field doesn't exist in the table. >>>> >>>> is it a typo ? or does it need a field concatenation of prefix and digits ? >>>> >>>> Thanks >>>> >>>> >>>> >>>> Franck >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/86dbe768/attachment-0001.html From msc at freeswitch.org Thu Jun 10 13:12:33 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Jun 2010 13:12:33 -0700 Subject: [Freeswitch-users] Weekly FreeSwitch Conference call June 9 @ 1pm EST In-Reply-To: References: Message-ID: On Thu, Jun 10, 2010 at 3:27 AM, Aza Tek wrote: > It seems a lot of the Weekly Conference Calls actually don't have > associated recordings, is this correct or am I just missing them? > I've seen some recordings in the Wiki and Conf Call pages, but for less > than half of all the conferences that are there. Should I also be looking > somewhere else? > We just recently started recording the conferences. We only specifically record and post on the wiki if there is a specific presentation. Are you looking for a specific presentation? Let us know. NormT and I (mercutioviz) will make sure that all the available recordings are up on the wiki. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/b0f47b84/attachment.html From msc at freeswitch.org Thu Jun 10 13:17:50 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Jun 2010 13:17:50 -0700 Subject: [Freeswitch-users] JavaScript control Transfer In-Reply-To: References: <760778166.8.1276124703365.JavaMail.root@srvr12.remotelinkml.com> <1276128865845-5161043.post@n2.nabble.com> Message-ID: On Thu, Jun 10, 2010 at 9:24 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > session.execute("transfer", ":'' inline"); > > Just to add, the inline dialplan trick is quite handy. More info here: http://wiki.freeswitch.org/wiki/Inline_Dialplan You can do crazy fun things with inline. Check out the cool auto-answer trick for portaudio calls. ;) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/0077cf62/attachment.html From helmut.kuper at ewetel.de Thu Jun 10 13:38:14 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 10 Jun 2010 22:38:14 +0200 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header Message-ID: <4C114D36.9000909@ewetel.de> Hello, I try to control the LEDs on Snom 370 phones using LUA and event type SEND_MESSAGE. The phone receives this message: MESSAGE sip:2850 at 85.16.245.213:1051 SIP/2.0 Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKeer7H9cFa85DN Max-Forwards: 70 From: ;tag=yH2p954XZZ1tj To: Call-ID: 2a6ed9ba-ef6d-122d-3d83-00144fe6e332 CSeq: 131979517 MESSAGE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-17097:17188M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/x-buttons Content-Length: 10 k=1 c=on and Snom acknowledges this with 200 OK. Unfortunately nothing happens. Snom's howto (http://wiki.snom.com/Category:HowTo:LED_Remote_Control) describes a message like this: MESSAGE sip:44 at 172.20.25.101:2048;transport=tls;line=44qsudyt SIP/2.0 Via: SIP/2.0/TLS 172.20.25.102:5061;branch=z9hG4bK-2909fe081f8466c9836c3b673f284842;rport From: ;tag=17820 To: Call-ID: umceg9jb at pbx CSeq: 35160 MESSAGE Max-Forwards: 70 Contact: Subject: buttons Content-Type: application/x-buttons Content-Length: 7 k=3 l=42 Main difference is the subject header containing "buttons" (despite of the body) Is there a way to add subject header as above to the message generated by FS? regards Helmut From brian at freeswitch.org Thu Jun 10 13:47:54 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Jun 2010 15:47:54 -0500 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: <4C114D36.9000909@ewetel.de> References: <4C114D36.9000909@ewetel.de> Message-ID: diff --git a/src/mod/endpoints/mod_sofia/mod_sofia.c b/src/mod/endpoints/mod_sofia/mod_sofia.c index 026ae42..1d274c2 100644 --- a/src/mod/endpoints/mod_sofia/mod_sofia.c +++ b/src/mod/endpoints/mod_sofia/mod_sofia.c @@ -4082,6 +4082,7 @@ static void general_event_handler(switch_event_t *event) const char *ct = switch_event_get_header(event, "content-type"); const char *user = switch_event_get_header(event, "user"); const char *host = switch_event_get_header(event, "host"); + const char *subject = switch_event_get_header(event, "subject"); const char *body = switch_event_get_body(event); sofia_profile_t *profile; nua_handle_t *nh; @@ -4114,7 +4115,10 @@ static void general_event_handler(switch_event_t *event) nh = nua_handle(profile->nua, NULL, NUTAG_URL(contact), SIPTAG_FROM_STR(id), SIPTAG_TO_STR(id), SIPTAG_CONTACT_STR(profile->url), TAG_END()); - nua_message(nh, NUTAG_NEWSUB(1), SIPTAG_CONTENT_TYPE_STR(ct), TAG_IF(!zstr(body), SIPTAG_PAYLOAD_STR(body)), TAG_END()); + nua_message(nh, NUTAG_NEWSUB(1), SIPTAG_CONTENT_TYPE_STR(ct), + TAG_IF(!zstr(body), SIPTAG_PAYLOAD_STR(body)), + TAG_IF(!zstr(subject), SIPTAG_SUBJECT_STR(subject)), + TAG_END()); free(id); This should work... please try it yourself. I'll push this later if that works. /b On Jun 10, 2010, at 3:38 PM, Helmut Kuper wrote: > Hello, > > I try to control the LEDs on Snom 370 phones using LUA and event type > SEND_MESSAGE. The phone receives this message: > > MESSAGE sip:2850 at 85.16.245.213:1051 SIP/2.0 > Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKeer7H9cFa85DN > Max-Forwards: 70 > From: ;tag=yH2p954XZZ1tj > To: > Call-ID: 2a6ed9ba-ef6d-122d-3d83-00144fe6e332 > CSeq: 131979517 MESSAGE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-17097:17188M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/x-buttons > Content-Length: 10 > > k=1 > c=on > > > > and Snom acknowledges this with 200 OK. Unfortunately nothing happens. > > Snom's howto (http://wiki.snom.com/Category:HowTo:LED_Remote_Control) > describes a message like this: > > MESSAGE sip:44 at 172.20.25.101:2048;transport=tls;line=44qsudyt SIP/2.0 > Via: SIP/2.0/TLS > 172.20.25.102:5061;branch=z9hG4bK-2909fe081f8466c9836c3b673f284842;rport > From: ;tag=17820 > To: > Call-ID: umceg9jb at pbx > CSeq: 35160 MESSAGE > Max-Forwards: 70 > Contact: > Subject: buttons > Content-Type: application/x-buttons > Content-Length: 7 > > k=3 > l=42 > > > Main difference is the subject header containing "buttons" (despite of > the body) > > Is there a way to add subject header as above to the message generated > by FS? > > regards > Helmut > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brian at freeswitch.org Thu Jun 10 13:49:07 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Jun 2010 15:49:07 -0500 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: <4C114D36.9000909@ewetel.de> References: <4C114D36.9000909@ewetel.de> Message-ID: Just git pull I just pushed it. /b On Jun 10, 2010, at 3:38 PM, Helmut Kuper wrote: > Main difference is the subject header containing "buttons" (despite of > the body) > > Is there a way to add subject header as above to the message generated > by FS? From anthony.minessale at gmail.com Thu Jun 10 13:58:50 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Jun 2010 15:58:50 -0500 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: References: <4C114D36.9000909@ewetel.de> Message-ID: if you are playing with buttons stuff you should look into mod_snom where it's abstracted more. On Thu, Jun 10, 2010 at 3:49 PM, Brian West wrote: > Just git pull I just pushed it. > > /b > > On Jun 10, 2010, at 3:38 PM, Helmut Kuper wrote: > > > Main difference is the subject header containing "buttons" (despite of > > the body) > > > > Is there a way to add subject header as above to the message generated > > by FS? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100610/b170cb9b/attachment.html From infos at madovsky.org Thu Jun 10 14:00:27 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 17:00:27 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: clear as beer. Thanks ! F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 4:10 PM Subject: Re: [Freeswitch-users] mod_lcr What I use that differs from default: digits | text | not null digits_prefix | prefix_range | not null Indexes: "idx_prefix" gist (digits_prefix gist_prefix_range_ops) I keep digits around so I can test both default behavior and new behavior. You only really NEED the second one. On insert I set them to the same value (eg: 12145551212). End is fine. Doesn't really matter. On Thu, Jun 10, 2010 at 12:44 PM, Madovsky wrote: Ok I understand now. but where to create digist_prefix in lcr table ? the end is ok ? Thanks F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 8:38 AM Subject: Re: [Freeswitch-users] mod_lcr Read the part about how to define the table + the gist index. The whole custom_sql thing assumes a familiarity with sql. you can choose to not have a digits_prefix column and just change the datatype of prefix to prefix. You can do what I did which is to have prefix be text and digits_prefix be of type prefix and a trigger to keep the two in sync. the key is that you are searching against the prefix column for which there is a GIST index. On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: ok thanks I will read again F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 3:09 AM Subject: Re: [Freeswitch-users] mod_lcr it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- -Rupa ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/f0013815/attachment-0001.html From brian at freeswitch.org Thu Jun 10 14:04:57 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Jun 2010 16:04:57 -0500 Subject: [Freeswitch-users] mod_lcr In-Reply-To: References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: Not sure what beer you drink but I haven't seen a clear beer.... sure you're not drinking wine? /b On Jun 10, 2010, at 4:00 PM, Madovsky wrote: > clear as beer. Thanks ! > > F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/e9ca0345/attachment.html From infos at madovsky.org Thu Jun 10 14:11:55 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 17:11:55 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: <3DB78C36B2DC49A3B13EB375EE045523@MOBILEE1705> No I stopped and now all seems to be more clear ;) ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 5:04 PM Subject: Re: [Freeswitch-users] mod_lcr Not sure what beer you drink but I haven't seen a clear beer.... sure you're not drinking wine? /b On Jun 10, 2010, at 4:00 PM, Madovsky wrote: clear as beer. Thanks ! F ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/e15be5e6/attachment.html From sos at sokhapkin.dyndns.org Thu Jun 10 14:21:32 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Thu, 10 Jun 2010 17:21:32 -0400 Subject: [Freeswitch-users] mod_lcr In-Reply-To: <3DB78C36B2DC49A3B13EB375EE045523@MOBILEE1705> References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> <3DB78C36B2DC49A3B13EB375EE045523@MOBILEE1705> Message-ID: <201006101721.32573.sos@sokhapkin.dyndns.org> No, it's impossible to get a clear understanding of something without a lot of drinking. Don't stop. On Thursday 10 June 2010, Madovsky wrote: > No I stopped and now all seems to be more clear ;) > ----- Original Message ----- > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Sent: Thursday, June 10, 2010 5:04 PM > Subject: Re: [Freeswitch-users] mod_lcr > > > Not sure what beer you drink but I haven't seen a clear beer.... sure > you're not drinking wine? > > > /b > > > On Jun 10, 2010, at 4:00 PM, Madovsky wrote: > > > clear as beer. Thanks ! > > F > > > > > --------------------------------------------------------------------------- > --- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Thu Jun 10 14:41:00 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Jun 2010 14:41:00 -0700 Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1276197658427-5164872.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> <1275859376110-5146705.post@n2.nabble.com> <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> <1275862175540-5146803.post@n2.nabble.com> <1276001266045-5153498.post@n2.nabble.com> <1276162496035-5162317.post@n2.nabble.com> <1276197658427-5164872.post@n2.nabble.com> Message-ID: > > Anyhow, thanks for the pointers and I'll take a look at Sofia and report > something to them if need be. I think my first option may be to disable > session timers on the test route initially, just to see if there's anything > else afoot. > Question: are they even willing to look at their SIP handling? Seriously, they are bombing out a call because they don't want to ignore a header that they feel is meaningless but that others feel isn't meaningless? If everyone writing a SIP stack did something capricious like drop a call because the packet had a valid header that they just didn't want to see at that point in the dialog then SIP interop would be royal mess of voodoo programming and vendor finger pointing. It's a good thing that silly SIP implementations are rare. Oh wait... -MC P.S. - As to digging into Sofia - you are either brave or foolhardy! ;) You might want to talk to the Sofia devs directly and ask them about this exact scenario. They might be able to save you some time, or at least guide you to the right spot in the code. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/d1067e52/attachment.html From jvera at foravatars.com Thu Jun 10 15:13:59 2010 From: jvera at foravatars.com (=?ISO-8859-1?Q?Jos=E9_Vera?=) Date: Thu, 10 Jun 2010 17:13:59 -0500 Subject: [Freeswitch-users] Help with Sofia Message-ID: Hi there, Again, I'm trying to get freeswitch working, It gave me credentials and the info needed to get the service working but when the user connects to it I'm getting this error: http://img821.imageshack.us/img821/4562/sofia.png , Anyone knows how to disable sofia AUTH or set it to allow anyone to use it easy? wich steps I need to do (will add this info to a guide I'm working on for non experienced users like me) :) Thanks in advance, Jorel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/ba6d3810/attachment.html From brian at freeswitch.org Thu Jun 10 15:27:06 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Jun 2010 17:27:06 -0500 Subject: [Freeswitch-users] Help with Sofia In-Reply-To: References: Message-ID: <1CFD470C-3E3A-4F45-9D00-1FD6EF7804BA@freeswitch.org> Did you try to read the comments in the sip_profile/internal.xml? Here are the options you're looking for: and remove this out: /b On Jun 10, 2010, at 5:13 PM, Jos? Vera wrote: > > Anyone knows how to disable sofia AUTH or set it to allow anyone to use it easy? wich steps I need to do (will add this info to a guide I'm working on for non experienced users like me) :) From sos at sokhapkin.dyndns.org Thu Jun 10 15:35:09 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Thu, 10 Jun 2010 18:35:09 -0400 Subject: [Freeswitch-users] Help with Sofia In-Reply-To: <1CFD470C-3E3A-4F45-9D00-1FD6EF7804BA@freeswitch.org> References: <1CFD470C-3E3A-4F45-9D00-1FD6EF7804BA@freeswitch.org> Message-ID: <201006101835.10266.sos@sokhapkin.dyndns.org> Why do you expect that somebody is reading samples? On Thursday 10 June 2010, Brian West wrote: > Did you try to read the comments in the sip_profile/internal.xml? > > Here are the options you're looking for: > > > > value="true"/> > > value="true"/> > > > > and remove this out: > > > > > /b > > On Jun 10, 2010, at 5:13 PM, Jos? Vera wrote: > > Anyone knows how to disable sofia AUTH or set it to allow anyone to use > > it easy? wich steps I need to do (will add this info to a guide I'm > > working on for non experienced users like me) :) > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brian at freeswitch.org Thu Jun 10 15:38:55 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Jun 2010 17:38:55 -0500 Subject: [Freeswitch-users] Help with Sofia In-Reply-To: <201006101835.10266.sos@sokhapkin.dyndns.org> References: <1CFD470C-3E3A-4F45-9D00-1FD6EF7804BA@freeswitch.org> <201006101835.10266.sos@sokhapkin.dyndns.org> Message-ID: Its the most logical place to look since most config files are commented. /b On Jun 10, 2010, at 5:35 PM, Sergey Okhapkin wrote: > Why do you expect that somebody is reading samples? > > On Thursday 10 June 2010, Brian West wrote: >> Did you try to read the comments in the sip_profile/internal.xml? >> >> Here are the options you're looking for: >> >> >> >> > value="true"/> >> >> > value="true"/> >> >> >> >> and remove this out: >> >> >> >> >> /b From jor3l at foravatars.com Thu Jun 10 16:28:03 2010 From: jor3l at foravatars.com (Jor3l Boa) Date: Thu, 10 Jun 2010 18:28:03 -0500 Subject: [Freeswitch-users] Help with Sofia In-Reply-To: References: <1CFD470C-3E3A-4F45-9D00-1FD6EF7804BA@freeswitch.org> <201006101835.10266.sos@sokhapkin.dyndns.org> Message-ID: Hi, thanks that seems to work :-) 2010/6/10 Brian West > Its the most logical place to look since most config files are commented. > > /b > > On Jun 10, 2010, at 5:35 PM, Sergey Okhapkin wrote: > > > Why do you expect that somebody is reading samples? > > > > On Thursday 10 June 2010, Brian West wrote: > >> Did you try to read the comments in the sip_profile/internal.xml? > >> > >> Here are the options you're looking for: > >> > >> > >> > >> >> value="true"/> > >> > >> >> value="true"/> > >> > >> > >> > >> and remove this out: > >> > >> > >> > >> > >> /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/ace30df9/attachment.html From fs-list at communicatefreely.net Thu Jun 10 17:04:19 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 10 Jun 2010 20:04:19 -0400 Subject: [Freeswitch-users] Freeswitch download link and configuration Setup webpage on website In-Reply-To: <113855.18830.qm@web43407.mail.sp1.yahoo.com> References: <113855.18830.qm@web43407.mail.sp1.yahoo.com> Message-ID: <4C117D83.50809@communicatefreely.net> I don't think anyone replied because the information you are looking for is posted in the wiki, with step by step instructions. It's laid out very well, with a how-to for each operating system. http://wiki.freeswitch.org/wiki/Installation_Guide Shaik basha wrote: > Is there anyone who can help me in this forum? As I have not seen any reply to my earlier mails. Your earliest response in this regard would be very much appreciated. Thanking you in advance. REgards, > > >> Good morning every one. I a m a new bie in freeswitch, >> though I tried to search the download page and configuration >> set up. but, I don't see any where. Can any one help me in >> this regard. I have spent several hours, though I was not >> succeeded. >> >> Hence I kindly request to let me know from where I can >> download and how to do configuration setup. Thanking in >> advance. earliest response in this regard would be very much >> appreciated. I would be very thankful and grateful for your >> kind information. Regards, >> >> shaikbashaatc >> +919246769086 >> >> >> >> > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 gcd at i.ph Thu Jun 10 19:20:18 2010 From: gcd at i.ph (Nandy Dagondon) Date: Fri, 11 Jun 2010 10:20:18 +0800 Subject: [Freeswitch-users] Freeswitch download link and configuration Setup webpage on website In-Reply-To: <4C117D83.50809@communicatefreely.net> References: <113855.18830.qm@web43407.mail.sp1.yahoo.com> <4C117D83.50809@communicatefreely.net> Message-ID: we can only lead a thirsty horse to the water. :-) On Fri, Jun 11, 2010 at 8:04 AM, Tim St. Pierre < fs-list at communicatefreely.net> wrote: > I don't think anyone replied because the information you are looking for is > posted in the wiki, with > step by step instructions. It's laid out very well, with a how-to for each > operating system. > > http://wiki.freeswitch.org/wiki/Installation_Guide > > > Shaik basha wrote: > > Is there anyone who can help me in this forum? As I have not seen any > reply to my earlier mails. Your earliest response in this regard would be > very much appreciated. Thanking you in advance. REgards, > > > > > >> Good morning every one. I a m a new bie in freeswitch, > >> though I tried to search the download page and configuration > >> set up. but, I don't see any where. Can any one help me in > >> this regard. I have spent several hours, though I was not > >> succeeded. > >> > >> Hence I kindly request to let me know from where I can > >> download and how to do configuration setup. Thanking in > >> advance. earliest response in this regard would be very much > >> appreciated. I would be very thankful and grateful for your > >> kind information. Regards, > >> > >> shaikbashaatc > >> +919246769086 > >> > >> > >> > >> > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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/20100611/d453d395/attachment.html From infos at madovsky.org Thu Jun 10 20:21:39 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 10 Jun 2010 23:21:39 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: I succeed to make mod_lcr works with prefix and postgresql 8.4.4 I have also changed the type of digits as varchar(20) because pg doesn't accept empty numeric field (unless you know a trick to do that). Thanks for your patience F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 4:10 PM Subject: Re: [Freeswitch-users] mod_lcr What I use that differs from default: digits | text | not null digits_prefix | prefix_range | not null Indexes: "idx_prefix" gist (digits_prefix gist_prefix_range_ops) I keep digits around so I can test both default behavior and new behavior. You only really NEED the second one. On insert I set them to the same value (eg: 12145551212). End is fine. Doesn't really matter. On Thu, Jun 10, 2010 at 12:44 PM, Madovsky wrote: Ok I understand now. but where to create digist_prefix in lcr table ? the end is ok ? Thanks F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 8:38 AM Subject: Re: [Freeswitch-users] mod_lcr Read the part about how to define the table + the gist index. The whole custom_sql thing assumes a familiarity with sql. you can choose to not have a digits_prefix column and just change the datatype of prefix to prefix. You can do what I did which is to have prefix be text and digits_prefix be of type prefix and a trigger to keep the two in sync. the key is that you are searching against the prefix column for which there is a GIST index. On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: ok thanks I will read again F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 3:09 AM Subject: Re: [Freeswitch-users] mod_lcr it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- -Rupa ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100610/10e468a6/attachment-0001.html From lakindia89 at gmail.com Thu Jun 10 22:43:07 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Fri, 11 Jun 2010 11:13:07 +0530 Subject: [Freeswitch-users] Receiving SEGV on Outbound ESL In-Reply-To: <5583C105-20E2-49DE-BE53-696BEF45B1FD@freeswitch.org> References: <5583C105-20E2-49DE-BE53-696BEF45B1FD@freeswitch.org> Message-ID: After updating to the current GIT the problem got solved. Many thanks. On Wed, Jun 9, 2010 at 7:31 PM, Brian West wrote: > Have you updated to the latest code in git? > > /b > > On Jun 9, 2010, at 7:06 AM, lakshmanan ganapathy wrote: > > > But as soon as the call comes to the Outbound ESL program, it receives > SIGSEGV. > > The child program prints the following: > > Before making a ESL connection 4 > > Got signal SEGV > > > > > > Can any one please tell me why this is happening?? > > > > And one more thing. If I execute the following API, it works fine as > expected. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/94b32349/attachment.html From engineerzuhairraza at gmail.com Thu Jun 10 23:22:39 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Fri, 11 Jun 2010 11:22:39 +0500 Subject: [Freeswitch-users] Query Mod_skypopen Message-ID: Hi Skype module cannot be loaded on Centos 5.5, Did anyone experienced it???? -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/c4a295a5/attachment.html From gmaruzz at celliax.org Thu Jun 10 23:39:55 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 11 Jun 2010 08:39:55 +0200 Subject: [Freeswitch-users] Query Mod_skypopen In-Reply-To: References: Message-ID: On Fri, Jun 11, 2010 at 8:22 AM, Zuhair Raza wrote: > Hi > Skype module cannot be loaded on Centos 5.5, Did anyone experienced it???? Zuhair, this is not true. Please refrain from such statements. YOU have problems (with linux and with skypopen), is not mod_skypopen that has problems. -giovanni > > -- > Regards, > Zuhair Raza > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From mustafa.pk at gmail.com Thu Jun 10 23:50:23 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Fri, 11 Jun 2010 11:50:23 +0500 Subject: [Freeswitch-users] Query Mod_skypopen In-Reply-To: References: Message-ID: Zuhair, Did you try to troubleshoot the issue? if yes please send us some debug logs. or at-least put some details in your next email e.g. what steps you have taken and where are you stuck at ? Best Regards. -m On Fri, Jun 11, 2010 at 11:22 AM, Zuhair Raza wrote: > Hi > Skype module cannot be loaded on Centos 5.5, Did anyone experienced it???? > > -- > Regards, > Zuhair Raza > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com From engineerzuhairraza at gmail.com Fri Jun 11 00:28:28 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Fri, 11 Jun 2010 12:28:28 +0500 Subject: [Freeswitch-users] Query Mod_skypopen In-Reply-To: References: Message-ID: Sorry sir It is my mistake I agree, I am a novoice user. I did "make sure" and it goes fine without any error. 2010-06-11 02:15:50.941138 [ERR] mod_skypopen.c:1209 rev [(nil)|37 ][ERROR 1209 ][none ][-1,-1,-1] open of skypopen.conf failed -ERR [module load file routine returned an error] 2010-06-11 02:15:50.941138 [CRIT] switch_loadable_module.c:882 Error Loading mule /usr/local/freeswitch/mod/mod_skypopen.so **Module load routine returned an error** sorry again and thanks for support On Fri, Jun 11, 2010 at 11:39 AM, Giovanni Maruzzelli wrote: > On Fri, Jun 11, 2010 at 8:22 AM, Zuhair Raza > wrote: > > Hi > > Skype module cannot be loaded on Centos 5.5, Did anyone experienced > it???? > > Zuhair, > > this is not true. > > Please refrain from such statements. > > YOU have problems (with linux and with skypopen), is not mod_skypopen > that has problems. > > -giovanni > > > > > -- > > Regards, > > Zuhair Raza > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/89119415/attachment.html From gmaruzz at celliax.org Fri Jun 11 01:20:02 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 11 Jun 2010 10:20:02 +0200 Subject: [Freeswitch-users] Query Mod_skypopen In-Reply-To: References: Message-ID: On Fri, Jun 11, 2010 at 9:28 AM, Zuhair Raza wrote: > 2010-06-11 02:15:50.941138 [ERR] mod_skypopen.c:1209 rev [(nil)|37 > ][ERROR 1209 ][none????? ][-1,-1,-1] open of skypopen.conf failed > > -ERR [module load file routine returned an error] Zuhair, you really have to make more attention! What you posted tell clearly: "open of skypopen.conf failed" You can read it yourself. This means: 1) you are not paying attention 2) you do not do your homework 3) you want others to do your homework You have not copied the config file where it belongs, as it is clearly written in the wiki page (and as is obvious), and as you would have found if you put a minimum of effort in what you are doing. -giovanni > > On Fri, Jun 11, 2010 at 11:39 AM, Giovanni Maruzzelli > wrote: >> >> On Fri, Jun 11, 2010 at 8:22 AM, Zuhair Raza >> wrote: >> > Hi >> > Skype module cannot be loaded on Centos 5.5, Did anyone experienced >> > it???? >> >> Zuhair, >> >> this is not true. >> >> Please refrain from such statements. >> >> YOU have problems (with linux and with skypopen), is not mod_skypopen >> that has problems. >> >> -giovanni >> >> > >> > -- >> > Regards, >> > Zuhair Raza >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Regards, > Zuhair Raza > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From azatek0 at gmail.com Fri Jun 11 03:02:29 2010 From: azatek0 at gmail.com (Aza Tek) Date: Fri, 11 Jun 2010 12:02:29 +0200 Subject: [Freeswitch-users] Git SubModule Add Message-ID: Hi, Is possible to only 'checkout' specific components of FreeSWITCH as submodules using Git? If not, will this be made possible in the near future? Thanks Aza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/141bb1e4/attachment.html From ashley at midletearth.com Fri Jun 11 03:03:39 2010 From: ashley at midletearth.com (Ashley B) Date: Fri, 11 Jun 2010 12:03:39 +0200 Subject: [Freeswitch-users] Change in behaviour for ESL events? In-Reply-To: <012001cb0662$cdbae390$6930aab0$@com> References: <1275896373992-5147891.post@n2.nabble.com> <012001cb0662$cdbae390$6930aab0$@com> Message-ID: <00a801cb094d$6049eb60$20ddc220$@com> How long would this problem have been occurring (I am using version 17048M for Windows)? Would it be resolved in the latest precompiled binaries for Windows? A From: Ashley B [mailto:ashley at midletearth.com] Sent: 07 June 2010 06:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Change in behaviour for ESL events? Hi Brian, Would this issue have affected not being to receive any "MESSAGE" events when binding to "all" events using EventConsumer in mod_managed? I receive every event BUT "MESSAGE" (afaik). Thanks Ashley From: Brian West [mailto:brian at freeswitch.org] Sent: 07 June 2010 04:58 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Change in behaviour for ESL events? You'll need to pull the source from git as this was fixed late last night. /b On Jun 7, 2010, at 2:39 AM, peely wrote: Hi, In the latest git snapshot I've compiled, event subscriptions within ESL no longer seem to function in the same way. I used to issue "filter Unique-ID {uuid}\n\n" and "filter Other-Leg-Unique-ID {uuid}\n\n" followed by "events plain all\n\n". I did this because "myevents\n\n" would not allow me to subscribe to events for background jobs issued by bgapi, which I do quite a lot. Applying the filter then subscribing to all events seemed the most stable and allowed me to subscribe to additional events should I need them. In the latest snapshot, I don't receive any events through this mechanism. I tried "event text all" as newly suggested on the ESL outbound wiki page, but this transmits a heap of white space to my socket then kills freeswitch! Could somebody please tell me if this is something that is "work in progress" and will ultimately resume old behaviour, or should I be doing something else to monitor events for my uuid and anything I spawn in that session? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/b5f974b2/attachment-0001.html From helmut.kuper at ewetel.de Fri Jun 11 04:11:41 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 11 Jun 2010 13:11:41 +0200 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: References: <4C114D36.9000909@ewetel.de> Message-ID: <4C1219ED.2040901@ewetel.de> Hi Brian, thx for your quick patch. I added it. Subject header is now there :) But still no success. Here the Snom log: Received from udp:85.16.246.6:5060 at 11/6/2010 13:09:58:371 (620 bytes): MESSAGE sip:2850 at 85.16.245.213:1066 SIP/2.0 Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK763Kg0N1grK6e Max-Forwards: 70 From: ;tag=BB5FgvDmS3BHa To: Call-ID: a49b9045-efec-122d-11ba-00144fe6e332 CSeq: 132008799 MESSAGE Contact: Subject: buttons User-Agent: FreeSWITCH-mod_sofia/1.0.head-17097:17188M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/x-buttons Content-Length: 11 k=11 c=on Sent to udp:85.16.246.6:5060 at 11/6/2010 13:09:58:455 (254 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 85.16.246.6;rport=5060;branch=z9hG4bK763Kg0N1grK6e From: ;tag=BB5FgvDmS3BHa To: Call-ID: a49b9045-efec-122d-11ba-00144fe6e332 CSeq: 132008799 MESSAGE Content-Length: 0 FKey 11 on Snom is configured as "button" Guess I have to ask Snom ... regrads Helmut On 10.06.2010 22:49, Brian West wrote: > Just git pull I just pushed it. > > /b > > On Jun 10, 2010, at 3:38 PM, Helmut Kuper wrote: > >> Main difference is the subject header containing "buttons" (despite of >> the body) >> >> Is there a way to add subject header as above to the message generated >> by FS? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 engineerzuhairraza at gmail.com Fri Jun 11 04:25:04 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Fri, 11 Jun 2010 16:25:04 +0500 Subject: [Freeswitch-users] Query Mod_skypopen In-Reply-To: References: Message-ID: Thank you sir I will pay full attention now and try to do everything by my own. It will be very useful for me. Thanks On Fri, Jun 11, 2010 at 1:20 PM, Giovanni Maruzzelli wrote: > On Fri, Jun 11, 2010 at 9:28 AM, Zuhair Raza > wrote: > > 2010-06-11 02:15:50.941138 [ERR] mod_skypopen.c:1209 rev [(nil)|37 > > ][ERROR 1209 ][none ][-1,-1,-1] open of skypopen.conf failed > > > > -ERR [module load file routine returned an error] > > Zuhair, you really have to make more attention! > > What you posted tell clearly: "open of skypopen.conf failed" > > You can read it yourself. > > This means: > 1) you are not paying attention > 2) you do not do your homework > 3) you want others to do your homework > > You have not copied the config file where it belongs, as it is clearly > written in the wiki page (and as is obvious), and as you would have > found if you put a minimum of effort in what you are doing. > > -giovanni > > > > > On Fri, Jun 11, 2010 at 11:39 AM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > > wrote: > >> > >> On Fri, Jun 11, 2010 at 8:22 AM, Zuhair Raza > >> wrote: > >> > Hi > >> > Skype module cannot be loaded on Centos 5.5, Did anyone experienced > >> > it???? > >> > >> Zuhair, > >> > >> this is not true. > >> > >> Please refrain from such statements. > >> > >> YOU have problems (with linux and with skypopen), is not mod_skypopen > >> that has problems. > >> > >> -giovanni > >> > >> > > >> > -- > >> > Regards, > >> > Zuhair Raza > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Regards, > > Zuhair Raza > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/aae1313a/attachment.html From rupa at rupa.com Fri Jun 11 04:56:54 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 11 Jun 2010 06:56:54 -0500 Subject: [Freeswitch-users] mod_lcr In-Reply-To: References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> <763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: null would be used for empty, but... why would you have an empty digits field? On Thu, Jun 10, 2010 at 10:21 PM, Madovsky wrote: > I succeed to make mod_lcr works with prefix and postgresql 8.4.4 > I have also changed the type of digits as varchar(20) because pg doesn't > accept empty numeric field (unless you know a trick to do that). > > Thanks for your patience > > F > > ----- Original Message ----- > *From:* Rupa Schomaker > *To:* freeswitch-users > *Sent:* Thursday, June 10, 2010 4:10 PM > *Subject:* Re: [Freeswitch-users] mod_lcr > > What I use that differs from default: > > digits | text | not null > digits_prefix | prefix_range | not null > Indexes: > "idx_prefix" gist (digits_prefix gist_prefix_range_ops) > > I keep digits around so I can test both default behavior and new behavior. > You only really NEED the second one. > > On insert I set them to the same value (eg: 12145551212). > > End is fine. Doesn't really matter. > > On Thu, Jun 10, 2010 at 12:44 PM, Madovsky wrote: > >> Ok I understand now. >> but where to create digist_prefix in lcr table ? the end is ok ? >> >> Thanks >> >> F >> >> ----- Original Message ----- >> *From:* Rupa Schomaker >> *To:* freeswitch-users >> *Sent:* Thursday, June 10, 2010 8:38 AM >> *Subject:* Re: [Freeswitch-users] mod_lcr >> >> Read the part about how to define the table + the gist index. The whole >> custom_sql thing assumes a familiarity with sql. you can choose to not have >> a digits_prefix column and just change the datatype of prefix to prefix. >> You can do what I did which is to have prefix be text and digits_prefix be >> of type prefix and a trigger to keep the two in sync. the key is that you >> are searching against the prefix column for which there is a GIST index. >> >> On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: >> >>> ok thanks I will read again >>> >>> F >>> >>> ----- Original Message ----- >>> *From:* Nandy Dagondon >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Sent:* Thursday, June 10, 2010 3:09 AM >>> *Subject:* Re: [Freeswitch-users] mod_lcr >>> >>> it's the digits_prefix in the WHERE clause that's causing the error. >>> >>> ur question re prefix+digits, it's explained in the Custom SQL portion in >>> the wiki. >>> >>> -nandy >>> >>> >>> On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: >>> >>>> ok so it needs also the alias l.digits in the condition I think. >>>> I'm a little confused about digits and prefix. >>>> if I check a number with the country code is it need to join >>>> prefix+digits ? how with this kinkd of sql request ? >>>> >>>> Thanks >>>> >>>> F >>>> >>>> ----- Original Message ----- >>>> *From:* Nandy Dagondon >>>> *To:* freeswitch-users at lists.freeswitch.org >>>> *Sent:* Thursday, June 10, 2010 2:44 AM >>>> *Subject:* Re: [Freeswitch-users] mod_lcr >>>> >>>> i think it's a typo. i changed digits_prefix to digits. to be sure, pls >>>> check the CREATE TABLE entries. >>>> -nandy >>>> >>>> >>>> On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: >>>> >>>>> I'm experimenting with mod_lcr with postgresql (8.4.4) >>>>> there s an example of custom sql on wiki below : >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> however the query failed cause of digits_prefix field doesn't exist in the table. >>>>> >>>>> is it a typo ? or does it need a field concatenation of prefix and digits ? >>>>> >>>>> Thanks >>>>> >>>>> >>>>> >>>>> Franck >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.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 >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> -Rupa >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> >> > > > -- > -Rupa > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/20ea0222/attachment-0001.html From nagalenoj at gmail.com Fri Jun 11 06:09:05 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Fri, 11 Jun 2010 18:39:05 +0530 Subject: [Freeswitch-users] leg_timeout isn't working. Message-ID: Dear friends, I've posted an issue in jira(2 days back) but I didn't get any response there. I just want to confirm whether any one else is facing the same and it is really an issue. Description: When I tried to execute 'bridge [leg_timeout=10]user/1010', it doesn't quit ringing if the callee didn't respond in 10 seconds. But when I use it in {leg_timeout=10}user/1010, it's working. When I refer the wiki, it is given as it shouldn't be used in curly braces. 'Can be used in per-leg [], but not in global {} for the dialstring.' -- From wiki (http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout) Attached the log here, http://jira.freeswitch.org/browse/MODAPP-433 -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/5a81897b/attachment.html From testeador01 at gmail.com Fri Jun 11 06:24:26 2010 From: testeador01 at gmail.com (Milena) Date: Fri, 11 Jun 2010 08:24:26 -0500 Subject: [Freeswitch-users] Freeswitch download link and configuration Setup webpage on website In-Reply-To: References: <113855.18830.qm@web43407.mail.sp1.yahoo.com> <4C117D83.50809@communicatefreely.net> Message-ID: Actually there were 2 replies on the original post: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-June/058782.html http://lists.freeswitch.org/pipermail/freeswitch-users/2010-June/058902.html Probably he wasn't waiting for them ;) 2010/6/10 Nandy Dagondon > we can only lead a thirsty horse to the water. :-) > > > > On Fri, Jun 11, 2010 at 8:04 AM, Tim St. Pierre < > fs-list at communicatefreely.net> wrote: > >> I don't think anyone replied because the information you are looking for >> is posted in the wiki, with >> step by step instructions. It's laid out very well, with a how-to for >> each operating system. >> >> http://wiki.freeswitch.org/wiki/Installation_Guide >> >> >> Shaik basha wrote: >> > Is there anyone who can help me in this forum? As I have not seen any >> reply to my earlier mails. Your earliest response in this regard would be >> very much appreciated. Thanking you in advance. REgards, >> > >> > >> >> Good morning every one. I a m a new bie in freeswitch, >> >> though I tried to search the download page and configuration >> >> set up. but, I don't see any where. Can any one help me in >> >> this regard. I have spent several hours, though I was not >> >> succeeded. >> >> >> >> Hence I kindly request to let me know from where I can >> >> download and how to do configuration setup. Thanking in >> >> advance. earliest response in this regard would be very much >> >> appreciated. I would be very thankful and grateful for your >> >> kind information. Regards, >> >> >> >> shaikbashaatc >> >> +919246769086 >> >> >> >> >> >> >> >> >> > >> > >> > >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.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 >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/5bf0858e/attachment.html From oliver.schenk at iinet.net.au Fri Jun 11 07:00:26 2010 From: oliver.schenk at iinet.net.au (Oliver Schenk) Date: Fri, 11 Jun 2010 22:00:26 +0800 Subject: [Freeswitch-users] FreeSwitch originate Message-ID: <4C12417A.3060503@iinet.net.au> Hi All, I'm making slow and steady progress through this Freeswitch world. I have a simple IVR running and I can listen to it when I call say extension 5000. I did this by testing with X-lite by registering as extension 1001. OS: Windows Version: 1.0.6 The question now is I can't make FreeSwitch make a call. What is the command? I tried: originate sofia/internal/1001 at 10.1.1.4 1000 This is the output I get: 2010-06-11 21:54:30.682927 [NOTICE] switch_channel.c:669 New Channel sofia/inter nal/1001 at 10.1.1.4 [e36dd3f5-a1ff-4c78-8e70-aca74533c068] 2010-06-11 21:54:30.686927 [NOTICE] switch_channel.c:669 New Channel sofia/exter nal/0000000000 at 10.1.1.4 [db5c5175-da5c-4568-bbad-59343222b399] 2010-06-11 21:54:30.688927 [INFO] mod_dialplan_xml.c:418 Processing ->1001 in co ntext public 2010-06-11 21:54:30.690928 [NOTICE] switch_ivr.c:1447 Transfer sofia/external/00 00000000 at 10.1.1.4 to XML[1001 at default] 2010-06-11 21:54:30.691928 [INFO] mod_dialplan_xml.c:418 Processing ->1001 in co ntext default 2010-06-11 21:54:30.693928 [INFO] mod_dptools.c:965 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/0000000000 at 10.1.1.4] Unique-ID: [db5c5175-da5c-4568-bbad-59343222b399] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [G7221] Channel-Read-Codec-Rate: [32000] Channel-Write-Codec-Name: [G7221] Channel-Write-Codec-Rate: [32000] Caller-Username: [0000000000] Caller-Dialplan: [XML] Caller-Caller-ID-Number: [0000000000] Caller-Network-Addr: [10.1.1.4] Caller-ANI: [0000000000] Caller-Destination-Number: [1001] Caller-Unique-ID: [db5c5175-da5c-4568-bbad-59343222b399] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-RDNIS: [1001] Caller-Channel-Name: [sofia/external/0000000000 at 10.1.1.4] Caller-Profile-Index: [2] Caller-Profile-Created-Time: [1276264470690928] Caller-Channel-Created-Time: [1276264470686927] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_uuid: [db5c5175-da5c-4568-bbad-59343222b399] variable_sip_local_network_addr: [10.1.1.4] variable_sip_network_ip: [10.1.1.4] variable_sip_network_port: [5070] variable_sip_received_ip: [10.1.1.4] variable_sip_received_port: [5070] variable_sip_via_protocol: [udp] variable_sip_from_user: [0000000000] variable_sip_from_uri: [0000000000 at 10.1.1.4] variable_sip_from_host: [10.1.1.4] variable_sip_from_user_stripped: [0000000000] variable_sip_from_tag: [1XeZvDa011mBe] variable_sofia_profile_name: [external] variable_sip_Remote-Party-ID: [;party=calling;screen=ye s;privacy=off] variable_sip_cid_type: [rpid] variable_sip_full_via: [SIP/2.0/UDP 10.1.1.4:5070;rport=5070;branch=z9hG4bKv0QyQ 4SSm612S] variable_sip_full_from: ["" ;tag=1XeZvDa011mBe] variable_sip_full_to: [] variable_sip_req_user: [1001] variable_sip_req_uri: [1001 at 10.1.1.4] variable_sip_req_host: [10.1.1.4] variable_sip_to_user: [1001] variable_sip_to_uri: [1001 at 10.1.1.4] variable_sip_to_host: [10.1.1.4] variable_sip_contact_user: [mod_sofia] variable_sip_contact_port: [5070] variable_sip_contact_uri: [mod_sofia at 10.1.1.4:5070] variable_sip_contact_host: [10.1.1.4] variable_channel_name: [sofia/external/0000000000 at 10.1.1.4] variable_sip_call_id: [b304f1fd-f003-122d-c789-5d32389d672f] variable_sip_user_agent: [FreeSWITCH-mod_sofia/1.0.6-exported] variable_sip_via_host: [10.1.1.4] variable_sip_via_port: [5070] variable_sip_via_rport: [5070] variable_switch_r_sdp: [v=0 o=FreeSWITCH 1276237680 1276237681 IN IP4 10.1.1.4 s=FreeSWITCH c=IN IP4 10.1.1.4 t=0 0 m=audio 26790 RTP/AVP 115 107 9 0 8 3 101 13 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ] variable_remote_media_ip: [10.1.1.4] variable_remote_media_port: [26790] variable_sip_use_codec_name: [G7221] variable_sip_use_codec_fmtp: [bitrate=48000] variable_sip_use_codec_rate: [32000] variable_sip_use_codec_ptime: [20] variable_read_codec: [G7221] variable_read_rate: [32000] variable_write_codec: [G7221] variable_write_rate: [32000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_max_forwards: [69] variable_current_application: [info] -ERR NO_USER_RESPONSE 2010-06-11 21:54:30.693928 [NOTICE] switch_core_state_machine.c:185 sofia/extern al/0000000000 at 10.1.1.4 has executed the last dialplan instruction, hanging up. 2010-06-11 21:54:30.693928 [NOTICE] switch_core_state_machine.c:187 Hangup sofia /external/0000000000 at 10.1.1.4 [CS_EXECUTE] [NORMAL_CLEARING] 2010-06-11 21:54:30.695928 [NOTICE] sofia.c:4789 Hangup sofia/internal/1001 at 10.1 .1.4 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2010-06-11 21:54:30.696928 [NOTICE] switch_core_session.c:1182 Session 6 (sofia/ external/0000000000 at 10.1.1.4) Ended 2010-06-11 21:54:30.696928 [NOTICE] switch_core_session.c:1184 Close Channel sof ia/external/0000000000 at 10.1.1.4 [CS_DESTROY] 2010-06-11 21:54:30.704928 [NOTICE] switch_core_session.c:1182 Session 5 (sofia/ internal/1001 at 10.1.1.4) Ended 2010-06-11 21:54:30.704928 [NOTICE] switch_core_session.c:1184 Close Channel sof ia/internal/1001 at 10.1.1.4 [CS_DESTROY] I somehow don't think I'm doing the right thing.... Pretty much all Freeswitch XML is still default except the password. Is Xlite allowed to be installed on the same computer as the freeswitch server? At this stage I'm not testing with any external gateways, just trying to get something working in my internal network. So should by domain be the IP of the server (10.1.1.4) or localhost? Do I need "mod_dialplan_directory? or just mod_dialplan_xml? **Confused** Thanks, Oliver From infos at madovsky.org Fri Jun 11 07:11:52 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 11 Jun 2010 10:11:52 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: I use prefix for country prefix and digits for npa-nxx digits. my trunk has a list with empty npa-nxx in some rows ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Friday, June 11, 2010 7:56 AM Subject: Re: [Freeswitch-users] mod_lcr null would be used for empty, but... why would you have an empty digits field? On Thu, Jun 10, 2010 at 10:21 PM, Madovsky wrote: I succeed to make mod_lcr works with prefix and postgresql 8.4.4 I have also changed the type of digits as varchar(20) because pg doesn't accept empty numeric field (unless you know a trick to do that). Thanks for your patience F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 4:10 PM Subject: Re: [Freeswitch-users] mod_lcr What I use that differs from default: digits | text | not null digits_prefix | prefix_range | not null Indexes: "idx_prefix" gist (digits_prefix gist_prefix_range_ops) I keep digits around so I can test both default behavior and new behavior. You only really NEED the second one. On insert I set them to the same value (eg: 12145551212). End is fine. Doesn't really matter. On Thu, Jun 10, 2010 at 12:44 PM, Madovsky wrote: Ok I understand now. but where to create digist_prefix in lcr table ? the end is ok ? Thanks F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 8:38 AM Subject: Re: [Freeswitch-users] mod_lcr Read the part about how to define the table + the gist index. The whole custom_sql thing assumes a familiarity with sql. you can choose to not have a digits_prefix column and just change the datatype of prefix to prefix. You can do what I did which is to have prefix be text and digits_prefix be of type prefix and a trigger to keep the two in sync. the key is that you are searching against the prefix column for which there is a GIST index. On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: ok thanks I will read again F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 3:09 AM Subject: Re: [Freeswitch-users] mod_lcr it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ---------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- -Rupa -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- -Rupa ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/0647256a/attachment-0001.html From helmut.kuper at ewetel.de Fri Jun 11 07:20:39 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 11 Jun 2010 16:20:39 +0200 Subject: [Freeswitch-users] no caller id name when using freeswitch.Session() in lua Message-ID: <4C124637.4010503@ewetel.de> Hello, I use this command to setup a session to a local phone: freeswitch.Session("{origination_caller_id_name=john,origination_caller_id_number=3}sofia/internal/2850 at 85.16.246.6") The display shows for caller's name/number "3/3" in stead of "john/3". So "origination_caller_id_name" seems to be ignored. Any ideas what I did wrong? I use FreeSWITCH Version 1.0.head (17097:17188M) regards Helmut From helmut.kuper at ewetel.de Fri Jun 11 07:34:18 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 11 Jun 2010 16:34:18 +0200 Subject: [Freeswitch-users] no caller id name when using freeswitch.Session() in lua In-Reply-To: <4C124637.4010503@ewetel.de> References: <4C124637.4010503@ewetel.de> Message-ID: <4C12496A.6080009@ewetel.de> Hi, sorry, my fault. Had an error in dialplan. regards Helmut From rupa at rupa.com Fri Jun 11 08:02:02 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 11 Jun 2010 10:02:02 -0500 Subject: [Freeswitch-users] mod_lcr In-Reply-To: References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> <763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: That doesn't sound right. prefix is for prepending the supplied number with data. So if you want to prepend the country code if someone just dials npa-nxx for instance. Usually it would be used to add the customer access code that some providers like to use. digits should be a full e.164 number without the +. So, US would be 1NPANXX for 1000s resolution. Normalize your numbers to e164 format prior to doing the lcr query. Using this methodology I can't see how you would have any empty digits fields. On Fri, Jun 11, 2010 at 9:11 AM, Madovsky wrote: > I use prefix for country prefix and digits for npa-nxx digits. > my trunk has a list with empty npa-nxx in some rows > > > > ----- Original Message ----- > *From:* Rupa Schomaker > *To:* freeswitch-users > *Sent:* Friday, June 11, 2010 7:56 AM > *Subject:* Re: [Freeswitch-users] mod_lcr > > null would be used for empty, but... why would you have an empty digits > field? > > On Thu, Jun 10, 2010 at 10:21 PM, Madovsky wrote: > >> I succeed to make mod_lcr works with prefix and postgresql 8.4.4 >> I have also changed the type of digits as varchar(20) because pg doesn't >> accept empty numeric field (unless you know a trick to do that). >> >> Thanks for your patience >> >> F >> >> ----- Original Message ----- >> *From:* Rupa Schomaker >> *To:* freeswitch-users >> *Sent:* Thursday, June 10, 2010 4:10 PM >> *Subject:* Re: [Freeswitch-users] mod_lcr >> >> What I use that differs from default: >> >> digits | text | not null >> digits_prefix | prefix_range | not null >> Indexes: >> "idx_prefix" gist (digits_prefix gist_prefix_range_ops) >> >> I keep digits around so I can test both default behavior and new >> behavior. You only really NEED the second one. >> >> On insert I set them to the same value (eg: 12145551212). >> >> End is fine. Doesn't really matter. >> >> On Thu, Jun 10, 2010 at 12:44 PM, Madovsky wrote: >> >>> Ok I understand now. >>> but where to create digist_prefix in lcr table ? the end is ok ? >>> >>> Thanks >>> >>> F >>> >>> ----- Original Message ----- >>> *From:* Rupa Schomaker >>> *To:* freeswitch-users >>> *Sent:* Thursday, June 10, 2010 8:38 AM >>> *Subject:* Re: [Freeswitch-users] mod_lcr >>> >>> Read the part about how to define the table + the gist index. The whole >>> custom_sql thing assumes a familiarity with sql. you can choose to not have >>> a digits_prefix column and just change the datatype of prefix to prefix. >>> You can do what I did which is to have prefix be text and digits_prefix be >>> of type prefix and a trigger to keep the two in sync. the key is that you >>> are searching against the prefix column for which there is a GIST index. >>> >>> On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: >>> >>>> ok thanks I will read again >>>> >>>> F >>>> >>>> ----- Original Message ----- >>>> *From:* Nandy Dagondon >>>> *To:* freeswitch-users at lists.freeswitch.org >>>> *Sent:* Thursday, June 10, 2010 3:09 AM >>>> *Subject:* Re: [Freeswitch-users] mod_lcr >>>> >>>> it's the digits_prefix in the WHERE clause that's causing the error. >>>> >>>> ur question re prefix+digits, it's explained in the Custom SQL portion >>>> in the wiki. >>>> >>>> -nandy >>>> >>>> >>>> On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: >>>> >>>>> ok so it needs also the alias l.digits in the condition I think. >>>>> I'm a little confused about digits and prefix. >>>>> if I check a number with the country code is it need to join >>>>> prefix+digits ? how with this kinkd of sql request ? >>>>> >>>>> Thanks >>>>> >>>>> F >>>>> >>>>> ----- Original Message ----- >>>>> *From:* Nandy Dagondon >>>>> *To:* freeswitch-users at lists.freeswitch.org >>>>> *Sent:* Thursday, June 10, 2010 2:44 AM >>>>> *Subject:* Re: [Freeswitch-users] mod_lcr >>>>> >>>>> i think it's a typo. i changed digits_prefix to digits. to be sure, pls >>>>> check the CREATE TABLE entries. >>>>> -nandy >>>>> >>>>> >>>>> On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: >>>>> >>>>>> I'm experimenting with mod_lcr with postgresql (8.4.4) >>>>>> there s an example of custom sql on wiki below : >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> however the query failed cause of digits_prefix field doesn't exist in the table. >>>>>> >>>>>> is it a typo ? or does it need a field concatenation of prefix and digits ? >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> >>>>>> Franck >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.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 >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.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 >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> -Rupa >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >>> >>> >> >> >> -- >> -Rupa >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> >> > > > -- > -Rupa > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/1abddf39/attachment-0001.html From bogdan.patrascoiu at sinergetic.ro Fri Jun 11 08:11:28 2010 From: bogdan.patrascoiu at sinergetic.ro (Bogdan Patrascoiu) Date: Fri, 11 Jun 2010 18:11:28 +0300 Subject: [Freeswitch-users] Post Hangup string In-Reply-To: <5E052784-C615-4E37-BE57-02856E54043C@gmail.com> References: <027FB72A-AD72-4A11-B0CA-790C13A34893@gmail.com> <5E052784-C615-4E37-BE57-02856E54043C@gmail.com> Message-ID: Hello, First thank you for the ideas, in this case the agent is leg A because it's an outbound callcenter the operators initiate all the calls. What we did in the dialplan was : ? ? ? ? ? ? ? ? ? The script for *3472 being : function mycb( session, type, obj, arg ) { try { if ( type == "dtmf" ) { console_log( "info", "digit: "+obj.digit+"\n" ); if ( obj.digit == "#" ) { //console_log( "info", "detected pound sign.\n" ); exit = true; return( false ); } dtmf.digits += obj.digit; if ( dtmf.digits.length >= digitmaxlength ) { exit = true; return( false ); } } } catch (e) { console_log( "err", e+"\n" ); } return( true ); var dtmf = new Object( ); dtmf.digits = ""; if ( session.ready( ) ) { session.answer( ); if (admin_pin.length > 0) { digitmaxlength = 2; session.streamFile( "/usr/local/freeswitch/sounds/custom/callcenter/1.wav", mycb, "dtmf"); session.collectInput( mycb, dtmf, timeoutpin ); //console_log( "info", "DISA pin: " + dtmf.digits + "\n" ); } while (dtmf.digits > 21 || dtmf.digits < 1) { session.streamFile( "/usr/local/freeswitch/sounds/custom/callcenter/2.wav", mycb, "dtmf"); dtmf.digits = ""; session.collectInput( mycb, dtmf, timeoutpin ); } session.streamFile( "/usr/local/freeswitch/sounds/custom/callcenter/3.wav", mycb, "dtmf"); session.execute("set","cod_status="+dtmf.digits); console_log( "info", "DISA Pin: " + dtmf.digits + " is incorrect\n" ); var record_file = session.getVariable("record_file"); session.hangup("NORMAL_CLEARING"); record_file = record_file.replace ( ".mp3",""); system("mv " + record_file + ".mp3 " + record_file + "_" + dtmf.digits + ".mp3"); console_log( "info", "Record file: " + record_file + "\n" ); Thanks again, Bogdan On Wed, Jun 9, 2010 at 7:32 PM, David Ponzone wrote: > Tony, > of course, you're right. > In my mind, it was an outbound call center, so the agent will be leg A. > But for an inbound call center, the agent would be leg B. > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 09/06/2010 ? 18:13, Anthony Minessale a ?crit : > > if you do that and the other end of the call hangs up first your application > will not execute. > FreeSWITCH does not continue in the?dial-plan?once a call has hungup. > you are better off using the api_hangup_hook variable to have a script be > called on hangup that processes the data. > The channel is not available in the script but there is an event with all of > the channel vars. > > On Wed, Jun 9, 2010 at 10:24 AM, David Ponzone > wrote: >> >> Bogdan, >> There could be a way to do something nice if you don't hangup leg A after >> the call. >> The idea would be to disable hangup_after_bridge (it is by default I >> think), and to add a call to an internal IVR after the call to the customer >> is completed. >> In this IVR, you would listen to a wrap-up code entered as DTMF, and then >> you would rename to recording of the first call according to this. >> Some work to do, but should be doable. >> David Ponzone ?Direction Technique >> email: david.ponzone at ipeva.fr >> tel: ? ? ?01 74 03 18 97 >> gsm: ? 06 66 98 76 34 >> Service Client?IPeva >> tel: ? ? ?0811 46 26 26 >> www.ipeva.fr? -? ?www.ipeva-studio.com >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce >> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. >> >> >> >> Le 08/06/2010 ? 23:12, Bogdan Patrascoiu a ?crit : >> >> Hello, >> >> I'm currently working on a FreeSWITCH box for a callcenter. One of the >> callcenter's requests was to rename each of the calls recordings based >> on the call's outcome, so each operator must tag each of it's call >> recording by hand. I planned to do this by ignoring both parties >> hangup requests and to catch a post end of conversation string from >> the voip client (in this case X-Lite) and fit this in the recording's >> file name by renaming it. >> >> I haven't done anything like it so far and wanted to ask if there is a >> standard approach for such post hangups implementations . >> >> >> Thanks, >> >> Bogdan Patrascoiu >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> > > > > -- > 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 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > From infos at madovsky.org Fri Jun 11 08:41:52 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 11 Jun 2010 11:41:52 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: Ha ok thanks. but the problem is usually trunk offer rates list with country cod and city/route in separated columns. so if I understand I need to modify all the lists everytime to join these to column in one ? F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Friday, June 11, 2010 11:02 AM Subject: Re: [Freeswitch-users] mod_lcr That doesn't sound right. prefix is for prepending the supplied number with data. So if you want to prepend the country code if someone just dials npa-nxx for instance. Usually it would be used to add the customer access code that some providers like to use. digits should be a full e.164 number without the +. So, US would be 1NPANXX for 1000s resolution. Normalize your numbers to e164 format prior to doing the lcr query. Using this methodology I can't see how you would have any empty digits fields. On Fri, Jun 11, 2010 at 9:11 AM, Madovsky wrote: I use prefix for country prefix and digits for npa-nxx digits. my trunk has a list with empty npa-nxx in some rows ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Friday, June 11, 2010 7:56 AM Subject: Re: [Freeswitch-users] mod_lcr null would be used for empty, but... why would you have an empty digits field? On Thu, Jun 10, 2010 at 10:21 PM, Madovsky wrote: I succeed to make mod_lcr works with prefix and postgresql 8.4.4 I have also changed the type of digits as varchar(20) because pg doesn't accept empty numeric field (unless you know a trick to do that). Thanks for your patience F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 4:10 PM Subject: Re: [Freeswitch-users] mod_lcr What I use that differs from default: digits | text | not null digits_prefix | prefix_range | not null Indexes: "idx_prefix" gist (digits_prefix gist_prefix_range_ops) I keep digits around so I can test both default behavior and new behavior. You only really NEED the second one. On insert I set them to the same value (eg: 12145551212). End is fine. Doesn't really matter. On Thu, Jun 10, 2010 at 12:44 PM, Madovsky wrote: Ok I understand now. but where to create digist_prefix in lcr table ? the end is ok ? Thanks F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 8:38 AM Subject: Re: [Freeswitch-users] mod_lcr Read the part about how to define the table + the gist index. The whole custom_sql thing assumes a familiarity with sql. you can choose to not have a digits_prefix column and just change the datatype of prefix to prefix. You can do what I did which is to have prefix be text and digits_prefix be of type prefix and a trigger to keep the two in sync. the key is that you are searching against the prefix column for which there is a GIST index. On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: ok thanks I will read again F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 3:09 AM Subject: Re: [Freeswitch-users] mod_lcr it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- -Rupa ---------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- -Rupa -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- -Rupa ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/be44f2a5/attachment-0001.html From rupa at rupa.com Fri Jun 11 09:06:18 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 11 Jun 2010 11:06:18 -0500 Subject: [Freeswitch-users] mod_lcr In-Reply-To: References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705> <763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: Yes, generally one must normalize the data on import as well. On Fri, Jun 11, 2010 at 10:41 AM, Madovsky wrote: > Ha ok thanks. but the problem is usually trunk offer rates list with > country cod and city/route in separated columns. > so if I understand I need to modify all the lists everytime to join these > to column in one ? > > F > > ----- Original Message ----- > *From:* Rupa Schomaker > *To:* freeswitch-users > *Sent:* Friday, June 11, 2010 11:02 AM > *Subject:* Re: [Freeswitch-users] mod_lcr > > That doesn't sound right. > > prefix is for prepending the supplied number with data. So if you want to > prepend the country code if someone just dials npa-nxx for instance. > Usually it would be used to add the customer access code that some > providers like to use. > > digits should be a full e.164 number without the +. So, US would be > 1NPANXX for 1000s resolution. > > Normalize your numbers to e164 format prior to doing the lcr query. > > Using this methodology I can't see how you would have any empty digits > fields. > > On Fri, Jun 11, 2010 at 9:11 AM, Madovsky wrote: > >> I use prefix for country prefix and digits for npa-nxx digits. >> my trunk has a list with empty npa-nxx in some rows >> >> >> >> ----- Original Message ----- >> *From:* Rupa Schomaker >> *To:* freeswitch-users >> *Sent:* Friday, June 11, 2010 7:56 AM >> *Subject:* Re: [Freeswitch-users] mod_lcr >> >> null would be used for empty, but... why would you have an empty digits >> field? >> >> On Thu, Jun 10, 2010 at 10:21 PM, Madovsky wrote: >> >>> I succeed to make mod_lcr works with prefix and postgresql 8.4.4 >>> I have also changed the type of digits as varchar(20) because pg doesn't >>> accept empty numeric field (unless you know a trick to do that). >>> >>> Thanks for your patience >>> >>> F >>> >>> ----- Original Message ----- >>> *From:* Rupa Schomaker >>> *To:* freeswitch-users >>> *Sent:* Thursday, June 10, 2010 4:10 PM >>> *Subject:* Re: [Freeswitch-users] mod_lcr >>> >>> What I use that differs from default: >>> >>> digits | text | not null >>> digits_prefix | prefix_range | not null >>> Indexes: >>> "idx_prefix" gist (digits_prefix gist_prefix_range_ops) >>> >>> I keep digits around so I can test both default behavior and new >>> behavior. You only really NEED the second one. >>> >>> On insert I set them to the same value (eg: 12145551212). >>> >>> End is fine. Doesn't really matter. >>> >>> On Thu, Jun 10, 2010 at 12:44 PM, Madovsky wrote: >>> >>>> Ok I understand now. >>>> but where to create digist_prefix in lcr table ? the end is ok ? >>>> >>>> Thanks >>>> >>>> F >>>> >>>> ----- Original Message ----- >>>> *From:* Rupa Schomaker >>>> *To:* freeswitch-users >>>> *Sent:* Thursday, June 10, 2010 8:38 AM >>>> *Subject:* Re: [Freeswitch-users] mod_lcr >>>> >>>> Read the part about how to define the table + the gist index. The whole >>>> custom_sql thing assumes a familiarity with sql. you can choose to not have >>>> a digits_prefix column and just change the datatype of prefix to prefix. >>>> You can do what I did which is to have prefix be text and digits_prefix be >>>> of type prefix and a trigger to keep the two in sync. the key is that you >>>> are searching against the prefix column for which there is a GIST index. >>>> >>>> On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: >>>> >>>>> ok thanks I will read again >>>>> >>>>> F >>>>> >>>>> ----- Original Message ----- >>>>> *From:* Nandy Dagondon >>>>> *To:* freeswitch-users at lists.freeswitch.org >>>>> *Sent:* Thursday, June 10, 2010 3:09 AM >>>>> *Subject:* Re: [Freeswitch-users] mod_lcr >>>>> >>>>> it's the digits_prefix in the WHERE clause that's causing the error. >>>>> >>>>> ur question re prefix+digits, it's explained in the Custom SQL portion >>>>> in the wiki. >>>>> >>>>> -nandy >>>>> >>>>> >>>>> On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: >>>>> >>>>>> ok so it needs also the alias l.digits in the condition I think. >>>>>> I'm a little confused about digits and prefix. >>>>>> if I check a number with the country code is it need to join >>>>>> prefix+digits ? how with this kinkd of sql request ? >>>>>> >>>>>> Thanks >>>>>> >>>>>> F >>>>>> >>>>>> ----- Original Message ----- >>>>>> *From:* Nandy Dagondon >>>>>> *To:* freeswitch-users at lists.freeswitch.org >>>>>> *Sent:* Thursday, June 10, 2010 2:44 AM >>>>>> *Subject:* Re: [Freeswitch-users] mod_lcr >>>>>> >>>>>> i think it's a typo. i changed digits_prefix to digits. to be sure, >>>>>> pls check the CREATE TABLE entries. >>>>>> -nandy >>>>>> >>>>>> >>>>>> On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: >>>>>> >>>>>>> I'm experimenting with mod_lcr with postgresql (8.4.4) >>>>>>> there s an example of custom sql on wiki below : >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> however the query failed cause of digits_prefix field doesn't exist in the table. >>>>>>> >>>>>>> is it a typo ? or does it need a field concatenation of prefix and digits ? >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> >>>>>>> >>>>>>> Franck >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.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 >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.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 >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> -Rupa >>>> >>>> ------------------------------ >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>>> >>>> >>> >>> >>> -- >>> -Rupa >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >>> >>> >> >> >> -- >> -Rupa >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> >> > > > -- > -Rupa > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/fb7643b3/attachment-0001.html From helmut.kuper at ewetel.de Fri Jun 11 09:19:40 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 11 Jun 2010 18:19:40 +0200 Subject: [Freeswitch-users] filter events in Lua Message-ID: <4C12621C.8070401@ewetel.de> Hello, Is there a way to filter events in Lua like this? filter Unique-ID regards Helmut From infos at madovsky.org Fri Jun 11 09:20:23 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 11 Jun 2010 12:20:23 -0400 Subject: [Freeswitch-users] mod_lcr References: <2455E92CF9674A1297A1665308C26346@MOBILEE1705><763E5182408E4F1C8A8E9EF5ABDBE887@MOBILEE1705> Message-ID: I see.... thanks for the tip. ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Friday, June 11, 2010 12:06 PM Subject: Re: [Freeswitch-users] mod_lcr Yes, generally one must normalize the data on import as well. On Fri, Jun 11, 2010 at 10:41 AM, Madovsky wrote: Ha ok thanks. but the problem is usually trunk offer rates list with country cod and city/route in separated columns. so if I understand I need to modify all the lists everytime to join these to column in one ? F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Friday, June 11, 2010 11:02 AM Subject: Re: [Freeswitch-users] mod_lcr That doesn't sound right. prefix is for prepending the supplied number with data. So if you want to prepend the country code if someone just dials npa-nxx for instance. Usually it would be used to add the customer access code that some providers like to use. digits should be a full e.164 number without the +. So, US would be 1NPANXX for 1000s resolution. Normalize your numbers to e164 format prior to doing the lcr query. Using this methodology I can't see how you would have any empty digits fields. On Fri, Jun 11, 2010 at 9:11 AM, Madovsky wrote: I use prefix for country prefix and digits for npa-nxx digits. my trunk has a list with empty npa-nxx in some rows ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Friday, June 11, 2010 7:56 AM Subject: Re: [Freeswitch-users] mod_lcr null would be used for empty, but... why would you have an empty digits field? On Thu, Jun 10, 2010 at 10:21 PM, Madovsky wrote: I succeed to make mod_lcr works with prefix and postgresql 8.4.4 I have also changed the type of digits as varchar(20) because pg doesn't accept empty numeric field (unless you know a trick to do that). Thanks for your patience F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 4:10 PM Subject: Re: [Freeswitch-users] mod_lcr What I use that differs from default: digits | text | not null digits_prefix | prefix_range | not null Indexes: "idx_prefix" gist (digits_prefix gist_prefix_range_ops) I keep digits around so I can test both default behavior and new behavior. You only really NEED the second one. On insert I set them to the same value (eg: 12145551212). End is fine. Doesn't really matter. On Thu, Jun 10, 2010 at 12:44 PM, Madovsky wrote: Ok I understand now. but where to create digist_prefix in lcr table ? the end is ok ? Thanks F ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users Sent: Thursday, June 10, 2010 8:38 AM Subject: Re: [Freeswitch-users] mod_lcr Read the part about how to define the table + the gist index. The whole custom_sql thing assumes a familiarity with sql. you can choose to not have a digits_prefix column and just change the datatype of prefix to prefix. You can do what I did which is to have prefix be text and digits_prefix be of type prefix and a trigger to keep the two in sync. the key is that you are searching against the prefix column for which there is a GIST index. On Thu, Jun 10, 2010 at 2:53 AM, Madovsky wrote: ok thanks I will read again F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 3:09 AM Subject: Re: [Freeswitch-users] mod_lcr it's the digits_prefix in the WHERE clause that's causing the error. ur question re prefix+digits, it's explained in the Custom SQL portion in the wiki. -nandy On Thu, Jun 10, 2010 at 2:56 PM, Madovsky wrote: ok so it needs also the alias l.digits in the condition I think. I'm a little confused about digits and prefix. if I check a number with the country code is it need to join prefix+digits ? how with this kinkd of sql request ? Thanks F ----- Original Message ----- From: Nandy Dagondon To: freeswitch-users at lists.freeswitch.org Sent: Thursday, June 10, 2010 2:44 AM Subject: Re: [Freeswitch-users] mod_lcr i think it's a typo. i changed digits_prefix to digits. to be sure, pls check the CREATE TABLE entries. -nandy On Thu, Jun 10, 2010 at 2:16 PM, Madovsky wrote: I'm experimenting with mod_lcr with postgresql (8.4.4) there s an example of custom sql on wiki below : however the query failed cause of digits_prefix field doesn't exist in the table.is it a typo ? or does it need a field concatenation of prefix and digits ?Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- -Rupa ------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- -Rupa ---------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/07242d8a/attachment-0001.html From anthony.minessale at gmail.com Fri Jun 11 09:25:56 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Jun 2010 11:25:56 -0500 Subject: [Freeswitch-users] leg_timeout isn't working. In-Reply-To: References: Message-ID: please expect more than a 2 day turnaround using free support on a free project. I have, in fact fixed your problem which was specific to using the user/ channel. please have more patience in the future. Once you open a jira someone will get to it eventually. On Fri, Jun 11, 2010 at 8:09 AM, Nagalenoj H. wrote: > Dear friends, > I've posted an issue in jira(2 days back) but I didn't get any response > there. I just want to confirm whether any one else is facing the same and it > is really an issue. > > Description: > When I tried to execute 'bridge [leg_timeout=10]user/1010', it doesn't quit > ringing if the callee didn't respond in 10 seconds. But when I use it in > {leg_timeout=10}user/1010, it's working. > When I refer the wiki, it is given as it shouldn't be used in curly braces. > > > 'Can be used in per-leg [], but not in global {} for the dialstring.' > -- From wiki ( > http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout) > > Attached the log here, > http://jira.freeswitch.org/browse/MODAPP-433 > > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100611/451007ca/attachment.html From anthony.minessale at gmail.com Fri Jun 11 09:29:40 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Jun 2010 11:29:40 -0500 Subject: [Freeswitch-users] FreeSwitch originate In-Reply-To: <4C12417A.3060503@iinet.net.au> References: <4C12417A.3060503@iinet.net.au> Message-ID: try originate user/1001 1000 On Fri, Jun 11, 2010 at 9:00 AM, Oliver Schenk wrote: > Hi All, > > I'm making slow and steady progress through this Freeswitch world. > I have a simple IVR running and I can listen to it when I call say > extension 5000. > I did this by testing with X-lite by registering as extension 1001. > > OS: Windows > Version: 1.0.6 > > The question now is I can't make FreeSwitch make a call. What is the > command? > I tried: > > originate sofia/internal/1001 at 10.1.1.4 1000 > > This is the output I get: > > 2010-06-11 21:54:30.682927 [NOTICE] switch_channel.c:669 New Channel > sofia/inter > nal/1001 at 10.1.1.4 [e36dd3f5-a1ff-4c78-8e70-aca74533c068] > 2010-06-11 21:54:30.686927 [NOTICE] switch_channel.c:669 New Channel > sofia/exter > nal/0000000000 at 10.1.1.4 [db5c5175-da5c-4568-bbad-59343222b399] > 2010-06-11 21:54:30.688927 [INFO] mod_dialplan_xml.c:418 Processing > ->1001 in co > ntext public > 2010-06-11 21:54:30.690928 [NOTICE] switch_ivr.c:1447 Transfer > sofia/external/00 > 00000000 at 10.1.1.4 to XML[1001 at default] > 2010-06-11 21:54:30.691928 [INFO] mod_dialplan_xml.c:418 Processing > ->1001 in co > ntext default > 2010-06-11 21:54:30.693928 [INFO] mod_dptools.c:965 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-State-Number: [4] > Channel-Name: [sofia/external/0000000000 at 10.1.1.4] > Unique-ID: [db5c5175-da5c-4568-bbad-59343222b399] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [ringing] > Channel-Read-Codec-Name: [G7221] > Channel-Read-Codec-Rate: [32000] > Channel-Write-Codec-Name: [G7221] > Channel-Write-Codec-Rate: [32000] > Caller-Username: [0000000000] > Caller-Dialplan: [XML] > Caller-Caller-ID-Number: [0000000000] > Caller-Network-Addr: [10.1.1.4] > Caller-ANI: [0000000000] > Caller-Destination-Number: [1001] > Caller-Unique-ID: [db5c5175-da5c-4568-bbad-59343222b399] > Caller-Source: [mod_sofia] > Caller-Context: [default] > Caller-RDNIS: [1001] > Caller-Channel-Name: [sofia/external/0000000000 at 10.1.1.4] > Caller-Profile-Index: [2] > Caller-Profile-Created-Time: [1276264470690928] > Caller-Channel-Created-Time: [1276264470686927] > Caller-Channel-Answered-Time: [0] > Caller-Channel-Progress-Time: [0] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > variable_uuid: [db5c5175-da5c-4568-bbad-59343222b399] > variable_sip_local_network_addr: [10.1.1.4] > variable_sip_network_ip: [10.1.1.4] > variable_sip_network_port: [5070] > variable_sip_received_ip: [10.1.1.4] > variable_sip_received_port: [5070] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [0000000000] > variable_sip_from_uri: [0000000000 at 10.1.1.4] > variable_sip_from_host: [10.1.1.4] > variable_sip_from_user_stripped: [0000000000] > variable_sip_from_tag: [1XeZvDa011mBe] > variable_sofia_profile_name: [external] > variable_sip_Remote-Party-ID: > [ > >;party=calling;screen=ye > s;privacy=off] > variable_sip_cid_type: [rpid] > variable_sip_full_via: [SIP/2.0/UDP > 10.1.1.4:5070;rport=5070;branch=z9hG4bKv0QyQ > 4SSm612S] > variable_sip_full_from: ["" > >;tag=1XeZvDa011mBe] > variable_sip_full_to: [>] > variable_sip_req_user: [1001] > variable_sip_req_uri: [1001 at 10.1.1.4] > variable_sip_req_host: [10.1.1.4] > variable_sip_to_user: [1001] > variable_sip_to_uri: [1001 at 10.1.1.4] > variable_sip_to_host: [10.1.1.4] > variable_sip_contact_user: [mod_sofia] > variable_sip_contact_port: [5070] > variable_sip_contact_uri: [mod_sofia at 10.1.1.4:5070] > variable_sip_contact_host: [10.1.1.4] > variable_channel_name: [sofia/external/0000000000 at 10.1.1.4] > variable_sip_call_id: [b304f1fd-f003-122d-c789-5d32389d672f] > variable_sip_user_agent: [FreeSWITCH-mod_sofia/1.0.6-exported] > variable_sip_via_host: [10.1.1.4] > variable_sip_via_port: [5070] > variable_sip_via_rport: [5070] > variable_switch_r_sdp: [v=0 > o=FreeSWITCH 1276237680 1276237681 IN IP4 10.1.1.4 > s=FreeSWITCH > c=IN IP4 10.1.1.4 > t=0 0 > m=audio 26790 RTP/AVP 115 107 9 0 8 3 101 13 > a=rtpmap:115 G7221/32000 > a=fmtp:115 bitrate=48000 > a=rtpmap:107 G7221/16000 > a=fmtp:107 bitrate=32000 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > ] > variable_remote_media_ip: [10.1.1.4] > variable_remote_media_port: [26790] > variable_sip_use_codec_name: [G7221] > variable_sip_use_codec_fmtp: [bitrate=48000] > variable_sip_use_codec_rate: [32000] > variable_sip_use_codec_ptime: [20] > variable_read_codec: [G7221] > variable_read_rate: [32000] > variable_write_codec: [G7221] > variable_write_rate: [32000] > variable_endpoint_disposition: [RECEIVED] > variable_outside_call: [true] > variable_max_forwards: [69] > variable_current_application: [info] > > > > -ERR NO_USER_RESPONSE > > 2010-06-11 21:54:30.693928 [NOTICE] switch_core_state_machine.c:185 > sofia/extern > al/0000000000 at 10.1.1.4 has executed the last dialplan instruction, > hanging up. > 2010-06-11 21:54:30.693928 [NOTICE] switch_core_state_machine.c:187 > Hangup sofia > /external/0000000000 at 10.1.1.4 [CS_EXECUTE] [NORMAL_CLEARING] > 2010-06-11 21:54:30.695928 [NOTICE] sofia.c:4789 Hangup > sofia/internal/1001 at 10.1 > .1.4 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > 2010-06-11 21:54:30.696928 [NOTICE] switch_core_session.c:1182 Session 6 > (sofia/ > external/0000000000 at 10.1.1.4) Ended > 2010-06-11 21:54:30.696928 [NOTICE] switch_core_session.c:1184 Close > Channel sof > ia/external/0000000000 at 10.1.1.4 [CS_DESTROY] > 2010-06-11 21:54:30.704928 [NOTICE] switch_core_session.c:1182 Session 5 > (sofia/ > internal/1001 at 10.1.1.4) Ended > 2010-06-11 21:54:30.704928 [NOTICE] switch_core_session.c:1184 Close > Channel sof > ia/internal/1001 at 10.1.1.4 [CS_DESTROY] > > > > > I somehow don't think I'm doing the right thing.... > Pretty much all Freeswitch XML is still default except the password. > > > Is Xlite allowed to be installed on the same computer as the freeswitch > server? > > At this stage I'm not testing with any external gateways, just trying to > get something working in my internal network. > So should by domain be the IP of the server (10.1.1.4) or localhost? > > > Do I need "mod_dialplan_directory? or just mod_dialplan_xml? > > > **Confused** > > > Thanks, > > > Oliver > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100611/24957d13/attachment.html From anthony.minessale at gmail.com Fri Jun 11 09:34:16 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Jun 2010 11:34:16 -0500 Subject: [Freeswitch-users] Change in behaviour for ESL events? In-Reply-To: <00a801cb094d$6049eb60$20ddc220$@com> References: <1275896373992-5147891.post@n2.nabble.com> <012001cb0662$cdbae390$6930aab0$@com> <00a801cb094d$6049eb60$20ddc220$@com> Message-ID: anybody using svn revisions should update. On Fri, Jun 11, 2010 at 5:03 AM, Ashley B wrote: > How long would this problem have been occurring (I am using version > 17048M for Windows)? Would it be resolved in the latest precompiled binaries > for Windows? > > > > A > > > > *From:* Ashley B [mailto:ashley at midletearth.com] > *Sent:* 07 June 2010 06:59 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Change in behaviour for ESL events? > > > > Hi Brian, > > > > Would this issue have affected not being to receive any ?MESSAGE? events > when binding to ?all? events using EventConsumer in mod_managed? I receive > every event BUT ?MESSAGE? (afaik). > > > > Thanks > > Ashley > > > > *From:* Brian West [mailto:brian at freeswitch.org] > *Sent:* 07 June 2010 04:58 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Change in behaviour for ESL events? > > > > You'll need to pull the source from git as this was fixed late last night. > > > > /b > > > > On Jun 7, 2010, at 2:39 AM, peely wrote: > > > > > Hi, > > In the latest git snapshot I've compiled, event subscriptions within ESL no > longer seem to function in the same way. > > I used to issue "filter Unique-ID {uuid}\n\n" and "filter > Other-Leg-Unique-ID {uuid}\n\n" followed by "events plain all\n\n". I did > this because "myevents\n\n" would not allow me to subscribe to events for > background jobs issued by bgapi, which I do quite a lot. Applying the > filter > then subscribing to all events seemed the most stable and allowed me to > subscribe to additional events should I need them. > > In the latest snapshot, I don't receive any events through this mechanism. > I > tried "event text all" as newly suggested on the ESL outbound wiki page, > but > this transmits a heap of white space to my socket then kills freeswitch! > > Could somebody please tell me if this is something that is "work in > progress" and will ultimately resume old behaviour, or should I be doing > something else to monitor events for my uuid and anything I spawn in that > session? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100611/502b0fc6/attachment-0001.html From helmut.kuper at ewetel.de Fri Jun 11 09:56:53 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 11 Jun 2010 18:56:53 +0200 Subject: [Freeswitch-users] Lua: Only one event type for EventConsumer()? Message-ID: <4C126AD5.70706@ewetel.de> Hello, I have some touble to subscribe to certain events in Lua. What I do is this: con = freeswitch.EventConsumer("channel_state channel_answer") same with this: con = freeswitch.EventConsumer("channel_state,channel_answer") and this: con = freeswitch.EventConsumer("channel_state|channel_answer") No events are received. When I do it in this way: con = freeswitch.EventConsumer("channel_state") it works, but I get only channel_state events. Using "all" helps, but it's way to much information to parse. Any way to do what I like to do? regards Helmut From kward at binarysignal.com Fri Jun 11 10:06:03 2010 From: kward at binarysignal.com (Kurt Ward) Date: Fri, 11 Jun 2010 10:06:03 -0700 Subject: [Freeswitch-users] Change in behaviour for ESL events? In-Reply-To: References: <1275896373992-5147891.post@n2.nabble.com> <012001cb0662$cdbae390$6930aab0$@com> <00a801cb094d$6049eb60$20ddc220$@com> Message-ID: <331434B6-59EF-46D5-9A96-711D0F55CF2D@binarysignal.com> The Windows build is broken on Git. The last rev I was able to build in VS 2008 Express, Win32 was SVN. On Jun 11, 2010, at 9:34 AM, Anthony Minessale wrote: > anybody using svn revisions should update. > > > On Fri, Jun 11, 2010 at 5:03 AM, Ashley B > wrote: > How long would this problem have been occurring (I am using version > 17048M for Windows)? Would it be resolved in the latest precompiled > binaries for Windows? > > > A > > > From: Ashley B [mailto:ashley at midletearth.com] > Sent: 07 June 2010 06:59 PM > > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Change in behaviour for ESL events? > > > Hi Brian, > > > Would this issue have affected not being to receive any ?MESSAGE? > events when binding to ?all? events using EventConsumer in > mod_managed? I receive every event BUT ?MESSAGE? (afaik). > > > Thanks > > Ashley > > > From: Brian West [mailto:brian at freeswitch.org] > Sent: 07 June 2010 04:58 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Change in behaviour for ESL events? > > > You'll need to pull the source from git as this was fixed late last > night. > > > /b > > > On Jun 7, 2010, at 2:39 AM, peely wrote: > > > > Hi, > > In the latest git snapshot I've compiled, event subscriptions within > ESL no > longer seem to function in the same way. > > I used to issue "filter Unique-ID {uuid}\n\n" and "filter > Other-Leg-Unique-ID {uuid}\n\n" followed by "events plain all\n\n". > I did > this because "myevents\n\n" would not allow me to subscribe to > events for > background jobs issued by bgapi, which I do quite a lot. Applying > the filter > then subscribing to all events seemed the most stable and allowed me > to > subscribe to additional events should I need them. > > In the latest snapshot, I don't receive any events through this > mechanism. I > tried "event text all" as newly suggested on the ESL outbound wiki > page, but > this transmits a heap of white space to my socket then kills > freeswitch! > > Could somebody please tell me if this is something that is "work in > progress" and will ultimately resume old behaviour, or should I be > doing > something else to monitor events for my uuid and anything I spawn in > that > session? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 mail at jankubr.com Thu Jun 10 15:07:11 2010 From: mail at jankubr.com (Jan Kubr) Date: Fri, 11 Jun 2010 00:07:11 +0200 Subject: [Freeswitch-users] DETECTED_SPEECH events while playing a file Message-ID: Hi, I'm trying to detect speech while playing a file over the event socket. I start detecting speech: call-command: execute execute-app-name: detect_speech execute-app-arg: pocketsphinx ivr_grammar ivr event-lock:true Then playback the file: call-command: execute execute-app-name: playback execute-app-arg: ivr.wav event-lock:false Then resume the speech detection: call-command: execute execute-app-name: detect_speech execute-app-arg: resume event-lock:true Now when I say something, I do see this in the FreeSWITCH console: 2010-06-10 23:29:06.994346 [DEBUG] mod_pocketsphinx.c:383 Recognized: ONE, Confidence: 100 But I don't receive the DETECTED_SPEECH events over the socket. However, if I wait until the file is played, that is until I see this in the FreeSWITCH console: 2010-06-10 20:48:50.512148 [DEBUG] switch_ivr_play_say.c:1428 done playing file and say something after that, I do get the DETECTED_SPEECH events. Is there a way I could receive these events while the file is being played? Thanks, Jan Kubr -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/4b266ff8/attachment.html From kdjakovic at hotmail.com Fri Jun 11 00:26:19 2010 From: kdjakovic at hotmail.com (katarina djakovic) Date: Fri, 11 Jun 2010 00:26:19 -0700 Subject: [Freeswitch-users] Missing CDRs with ATTENDED TRANSFER Message-ID: Dear Freeswitch users, we have a problem where no CDRs are created for assisted transfer. We are running FS 1.0.6. We have extensions and one gateway to connect to PSTN. The scenario is the following: 1) Extension A calls a PSTN number (through the gateway) lets say nubmer C. Extension A is using Twinkle soft phone. Connection gets established. 2) Then, using Twinkle, extension A chooses to transfer the call to extension B by pressing xfer button in Twinkle's GUI. A calls B. 3) Then A hangs up (presses xfer button again) while connection gets established between B and C. However, FS produces only 2 CDRs for the whole process. Namely, there is a CDR for the step 1) A -> C. There is a CDR for 2) A -> B, but there is no CDR for 3) B -> C. The two CDRs are produced just after the call is transfered, but the third CDR never gets produced. Although, the FS is aware of the ATTENDED_TRANSFER as this appears as the 'reason' in the second CDR. Here are the CDRs produced: "2000","2000","gateway","external",""Katarina Piljanovic" <2000>","sofia/external/2000 at 95.180.91.118:5080","sofia/external/381117543472","bridge","sofia/g ateway/provider/381117543472","2010-06-10 16:52:11","2010-06-10 16:52:17","2010-06-10 16:55:58","227","221","NORMAL_CLEARING","","eaf26c9d-4cd7-4f65-a53e- a426b50da52d","","","GSM","GSM","none","none" "2000","2000","2002","external",""Katarina Piljanovic" <2000>","sofia/external/2000 at 95.180.91.118:5080","sofia/external/2002","bridge","{disable_rtp_au to_adjust=true}sofia/external/2002%95.180.91.118","2010-06-10 16:53:31","2010-06-10 16:53:33","2010-06-10 16:56:00","149","147","ATTENDED_TRANSFER","","ae6547cb-983d-405b-a4c1- 1afcd3260c4d","","","GSM","GSM","none","none" Is there a way to get the third CDR, is it something in config that we should set? As, this way the B and C can talk forever and never get billed for the call. Do you need to see the Dialplan to suggest the resolution? Also, the first CDR is not correct either, as it should list the number called instead of the word "gateway" in the third coloumn, as it normally does when we call through the gateway. Many thanks in advance, Katarina _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/7cf2f239/attachment.html From anthony.minessale at gmail.com Fri Jun 11 10:15:34 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Jun 2010 12:15:34 -0500 Subject: [Freeswitch-users] Change in behaviour for ESL events? In-Reply-To: <331434B6-59EF-46D5-9A96-711D0F55CF2D@binarysignal.com> References: <1275896373992-5147891.post@n2.nabble.com> <012001cb0662$cdbae390$6930aab0$@com> <00a801cb094d$6049eb60$20ddc220$@com> <331434B6-59EF-46D5-9A96-711D0F55CF2D@binarysignal.com> Message-ID: If that is the case then you should have reported that.... Most likely you missed some of the instructions for building for windows with git because nobody else has reported it. On Fri, Jun 11, 2010 at 12:06 PM, Kurt Ward wrote: > The Windows build is broken on Git. The last rev I was able to build > in VS 2008 Express, Win32 was SVN. > > On Jun 11, 2010, at 9:34 AM, Anthony Minessale wrote: > > > anybody using svn revisions should update. > > > > > > On Fri, Jun 11, 2010 at 5:03 AM, Ashley B > > wrote: > > How long would this problem have been occurring (I am using version > > 17048M for Windows)? Would it be resolved in the latest precompiled > > binaries for Windows? > > > > > > A > > > > > > From: Ashley B [mailto:ashley at midletearth.com] > > Sent: 07 June 2010 06:59 PM > > > > > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Change in behaviour for ESL events? > > > > > > Hi Brian, > > > > > > Would this issue have affected not being to receive any ?MESSAGE? > > events when binding to ?all? events using EventConsumer in > > mod_managed? I receive every event BUT ?MESSAGE? (afaik). > > > > > > Thanks > > > > Ashley > > > > > > From: Brian West [mailto:brian at freeswitch.org] > > Sent: 07 June 2010 04:58 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Change in behaviour for ESL events? > > > > > > You'll need to pull the source from git as this was fixed late last > > night. > > > > > > /b > > > > > > On Jun 7, 2010, at 2:39 AM, peely wrote: > > > > > > > > Hi, > > > > In the latest git snapshot I've compiled, event subscriptions within > > ESL no > > longer seem to function in the same way. > > > > I used to issue "filter Unique-ID {uuid}\n\n" and "filter > > Other-Leg-Unique-ID {uuid}\n\n" followed by "events plain all\n\n". > > I did > > this because "myevents\n\n" would not allow me to subscribe to > > events for > > background jobs issued by bgapi, which I do quite a lot. Applying > > the filter > > then subscribing to all events seemed the most stable and allowed me > > to > > subscribe to additional events should I need them. > > > > In the latest snapshot, I don't receive any events through this > > mechanism. I > > tried "event text all" as newly suggested on the ESL outbound wiki > > page, but > > this transmits a heap of white space to my socket then kills > > freeswitch! > > > > Could somebody please tell me if this is something that is "work in > > progress" and will ultimately resume old behaviour, or should I be > > doing > > something else to monitor events for my uuid and anything I spawn in > > that > > session? > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > -- 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/20100611/09ad621c/attachment-0001.html From anthony.minessale at gmail.com Fri Jun 11 10:18:01 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Jun 2010 12:18:01 -0500 Subject: [Freeswitch-users] DETECTED_SPEECH events while playing a file In-Reply-To: References: Message-ID: are you using sync or async mode on the socket. On Thu, Jun 10, 2010 at 5:07 PM, Jan Kubr wrote: > Hi, > I'm trying to detect speech while playing a file over the event socket. > > I start detecting speech: > > call-command: execute > execute-app-name: detect_speech > execute-app-arg: pocketsphinx ivr_grammar ivr > event-lock:true > > Then playback the file: > > call-command: execute > execute-app-name: playback > execute-app-arg: ivr.wav > event-lock:false > > Then resume the speech detection: > > call-command: execute > execute-app-name: detect_speech > execute-app-arg: resume > event-lock:true > > > Now when I say something, I do see this in the FreeSWITCH console: > > 2010-06-10 23:29:06.994346 [DEBUG] mod_pocketsphinx.c:383 Recognized: ONE, > Confidence: 100 > > But I don't receive the DETECTED_SPEECH events over the socket. > > However, if I wait until the file is played, that is until I see this in > the FreeSWITCH console: > > 2010-06-10 20:48:50.512148 [DEBUG] switch_ivr_play_say.c:1428 done playing > file > > and say something after that, I do get the DETECTED_SPEECH events. > > > Is there a way I could receive these events while the file is being played? > > Thanks, > Jan Kubr > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100611/664e23dd/attachment.html From mike at jerris.com Fri Jun 11 10:32:31 2010 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Jun 2010 13:32:31 -0400 Subject: [Freeswitch-users] Git SubModule Add In-Reply-To: References: Message-ID: <5767EFCF-8129-4DD4-90C7-0A50CEC974F3@jerris.com> You can do a sparse checkout with newer git. http://vmiklos.hu/blog/sparse-checkout-example-in-git-1-7 Mike On Jun 11, 2010, at 6:02 AM, Aza Tek wrote: > Hi, > > Is possible to only 'checkout' specific components of FreeSWITCH as submodules using Git? > If not, will this be made possible in the near future? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/6d72c6ae/attachment.html From peder at networkoblivion.com Fri Jun 11 11:22:36 2010 From: peder at networkoblivion.com (Peder) Date: Fri, 11 Jun 2010 13:22:36 -0500 Subject: [Freeswitch-users] Missing CDRs with ATTENDED TRANSFER In-Reply-To: References: Message-ID: <024a01cb0993$1194d480$34be7d80$@com> What value do you have for "legs" in cdr_csv.conf.xml? This is just a guess, but you probably need to have AB to get the third cdr. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of katarina djakovic Sent: Friday, June 11, 2010 2:26 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Missing CDRs with ATTENDED TRANSFER Dear Freeswitch users, we have a problem where no CDRs are created for assisted transfer. We are running FS 1.0.6. We have extensions and one gateway to connect to PSTN. The scenario is the following: 1) Extension A calls a PSTN number (through the gateway) lets say nubmer C. Extension A is using Twinkle soft phone. Connection gets established. 2) Then, using Twinkle, extension A chooses to transfer the call to extension B by pressing xfer button in Twinkle's GUI. A calls B. 3) Then A hangs up (presses xfer button again) while connection gets established between B and C. However, FS produces only 2 CDRs for the whole process. Namely, there is a CDR for the step 1) A -> C. There is a CDR for 2) A -> B, but there is no CDR for 3) B -> C. The two CDRs are produced just after the call is transfered, but the third CDR never gets produced. Although, the FS is aware of the ATTENDED_TRANSFER as this appears as the 'reason' in the second CDR. Here are the CDRs produced: "2000","2000","gateway","external",""Katarina Piljanovic" <2000>","sofia/external/2000 at 95.180.91.118:5080 ","sofia/external/381117543472","bridge","sofia/g ateway/provider/381117543472","2010-06-10 16:52:11","2010-06-10 16:52:17","2010-06-10 16:55:58","227","221","NORMAL_CLEARING","","eaf26c9d-4cd7-4f65-a53e- a426b50da52d","","","GSM","GSM","none","none" "2000","2000","2002","external",""Katarina Piljanovic" <2000>","sofia/external/2000 at 95.180.91.118:5080 ","sofia/external/2002","bridge","{disable_rtp_au to_adjust=true}sofia/external/2002%95.180.91.118","2010-06-10 16:53:31","2010-06-10 16:53:33","2010-06-10 16:56:00","149","147","ATTENDED_TRANSFER","","ae6547cb-983d-405b-a4c1- 1afcd3260c4d","","","GSM","GSM","none","none" Is there a way to get the third CDR, is it something in config that we should set? As, this way the B and C can talk forever and never get billed for the call. Do you need to see the Dialplan to suggest the resolution? Also, the first CDR is not correct either, as it should list the number called instead of the word "gateway" in the third coloumn, as it normally does when we call through the gateway. Many thanks in advance, Katarina _____ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. Learn more. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/540e9038/attachment.html From anthony.minessale at gmail.com Fri Jun 11 11:25:49 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Jun 2010 13:25:49 -0500 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: <4C1219ED.2040901@ewetel.de> References: <4C114D36.9000909@ewetel.de> <4C1219ED.2040901@ewetel.de> Message-ID: read my last post On Fri, Jun 11, 2010 at 6:11 AM, Helmut Kuper wrote: > Hi Brian, > > thx for your quick patch. > > I added it. Subject header is now there :) But still no success. > > Here the Snom log: > > Received from udp:85.16.246.6:5060 at 11/6/2010 13:09:58:371 (620 bytes): > > MESSAGE sip:2850 at 85.16.245.213:1066 SIP/2.0 > Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK763Kg0N1grK6e > Max-Forwards: 70 > From: >;tag=BB5FgvDmS3BHa > To: > > Call-ID: a49b9045-efec-122d-11ba-00144fe6e332 > CSeq: 132008799 MESSAGE > Contact: > Subject: buttons > User-Agent: FreeSWITCH-mod_sofia/1.0.head-17097:17188M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/x-buttons > Content-Length: 11 > > k=11 > c=on > > Sent to udp:85.16.246.6:5060 at 11/6/2010 13:09:58:455 (254 bytes): > > SIP/2.0 200 Ok > Via: SIP/2.0/UDP 85.16.246.6;rport=5060;branch=z9hG4bK763Kg0N1grK6e > From: >;tag=BB5FgvDmS3BHa > To: > > Call-ID: a49b9045-efec-122d-11ba-00144fe6e332 > CSeq: 132008799 MESSAGE > Content-Length: 0 > > > > FKey 11 on Snom is configured as "button" > > > Guess I have to ask Snom ... > > > regrads > Helmut > > > On 10.06.2010 22:49, Brian West wrote: > > Just git pull I just pushed it. > > > > /b > > > > On Jun 10, 2010, at 3:38 PM, Helmut Kuper wrote: > > > >> Main difference is the subject header containing "buttons" (despite of > >> the body) > >> > >> Is there a way to add subject header as above to the message generated > >> by FS? > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > -- 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/20100611/e076c88c/attachment-0001.html From anthony.minessale at gmail.com Fri Jun 11 11:30:10 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Jun 2010 13:30:10 -0500 Subject: [Freeswitch-users] filter events in Lua In-Reply-To: <4C12621C.8070401@ewetel.de> References: <4C12621C.8070401@ewetel.de> Message-ID: lua for ESL? On Fri, Jun 11, 2010 at 11:19 AM, Helmut Kuper wrote: > Hello, > > Is there a way to filter events in Lua like this? > > filter Unique-ID > > > regards > Helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100611/edc9f539/attachment.html From helmut.kuper at ewetel.de Fri Jun 11 11:54:16 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 11 Jun 2010 20:54:16 +0200 Subject: [Freeswitch-users] filter events in Lua In-Reply-To: References: <4C12621C.8070401@ewetel.de> Message-ID: <4C128658.6080808@ewetel.de> Hi Anthony, no, screnario is this: - Call comes in - FS dialplan calls lua script - lua script is starting bgapi command (originate) - lua script starts EventConsumer("all") - lua script is waiting for events I want to get only those events sent by the channel created by bgapi originate ... On 11.06.2010 20:30, Anthony Minessale wrote: > lua for ESL? > > On Fri, Jun 11, 2010 at 11:19 AM, Helmut Kuper > wrote: > > Hello, > > Is there a way to filter events in Lua like this? > > filter Unique-ID > regards Helmut From msc at freeswitch.org Fri Jun 11 12:34:27 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Jun 2010 12:34:27 -0700 Subject: [Freeswitch-users] Missing CDRs with ATTENDED TRANSFER In-Reply-To: <024a01cb0993$1194d480$34be7d80$@com> References: <024a01cb0993$1194d480$34be7d80$@com> Message-ID: On Fri, Jun 11, 2010 at 11:22 AM, Peder wrote: > What value do you have for ?legs? in cdr_csv.conf.xml? This is just a > guess, but you probably need to have AB to get the third cdr. > > > > > > > Just be warned that you probably will end up with *four* CDRs if you turn on the b-leg logging. I've tested these scenarios with Polycoms and Snoms and it is pretty consistent. Attended transfers are a challenge for CDRs but once you see the b-legs' CDRs you'll start to grasp it. As an exercise, turn on the b-leg logging and then do some experiments. For instance, watch your CDRs at these points during the call flow: After A dials B and B answers but *before* A actually transfers the call to B After A transfer the call to B but before B and C disconnect After B and C disconnect When you see four CDRs for what is essentially a single phone call it helps to know at what stage in the call flow that each of the records was created. Have fun. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/8b6c7d14/attachment.html From anthony.minessale at gmail.com Fri Jun 11 12:57:55 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Jun 2010 14:57:55 -0500 Subject: [Freeswitch-users] filter events in Lua In-Reply-To: <4C128658.6080808@ewetel.de> References: <4C12621C.8070401@ewetel.de> <4C128658.6080808@ewetel.de> Message-ID: you can just filter them yourself in your script. the filter you referenced is part of event socket not the core event system so you would have to implement it yourself. On Fri, Jun 11, 2010 at 1:54 PM, Helmut Kuper wrote: > Hi Anthony, > > no, screnario is this: > > - Call comes in > - FS dialplan calls lua script > - lua script is starting bgapi command (originate) > - lua script starts EventConsumer("all") > - lua script is waiting for events > > I want to get only those events sent by the channel created by bgapi > originate ... > > > > On 11.06.2010 20:30, Anthony Minessale wrote: > > lua for ESL? > > > > On Fri, Jun 11, 2010 at 11:19 AM, Helmut Kuper > > wrote: > > > > Hello, > > > > Is there a way to filter events in Lua like this? > > > > filter Unique-ID > > > > regards > Helmut > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100611/90cd7bd8/attachment.html From peter.olsson at visionutveckling.se Fri Jun 11 13:34:34 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 11 Jun 2010 22:34:34 +0200 Subject: [Freeswitch-users] Change in behaviour for ESL events? In-Reply-To: References: <1275896373992-5147891.post@n2.nabble.com> <012001cb0662$cdbae390$6930aab0$@com> <00a801cb094d$6049eb60$20ddc220$@com> <331434B6-59EF-46D5-9A96-711D0F55CF2D@binarysignal.com>, Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C567CA60751@cooper> I build in Windows almost every day - it works just fine. It's probably the old "CRFL-trick" that's ithe cause of the problems... /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Anthony Minessale [anthony.minessale at gmail.com] Skickat: den 11 juni 2010 19:15 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Change in behaviour for ESL events? If that is the case then you should have reported that.... Most likely you missed some of the instructions for building for windows with git because nobody else has reported it. On Fri, Jun 11, 2010 at 12:06 PM, Kurt Ward > wrote: The Windows build is broken on Git. The last rev I was able to build in VS 2008 Express, Win32 was SVN. On Jun 11, 2010, at 9:34 AM, Anthony Minessale wrote: > anybody using svn revisions should update. > > > On Fri, Jun 11, 2010 at 5:03 AM, Ashley B > > wrote: > How long would this problem have been occurring (I am using version > 17048M for Windows)? Would it be resolved in the latest precompiled > binaries for Windows? > > > A > > > From: Ashley B [mailto:ashley at midletearth.com] > Sent: 07 June 2010 06:59 PM > > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Change in behaviour for ESL events? > > > Hi Brian, > > > Would this issue have affected not being to receive any ?MESSAGE? > events when binding to ?all? events using EventConsumer in > mod_managed? I receive every event BUT ?MESSAGE? (afaik). > > > Thanks > > Ashley > > > From: Brian West [mailto:brian at freeswitch.org] > Sent: 07 June 2010 04:58 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Change in behaviour for ESL events? > > > You'll need to pull the source from git as this was fixed late last > night. > > > /b > > > On Jun 7, 2010, at 2:39 AM, peely wrote: > > > > Hi, > > In the latest git snapshot I've compiled, event subscriptions within > ESL no > longer seem to function in the same way. > > I used to issue "filter Unique-ID {uuid}\n\n" and "filter > Other-Leg-Unique-ID {uuid}\n\n" followed by "events plain all\n\n". > I did > this because "myevents\n\n" would not allow me to subscribe to > events for > background jobs issued by bgapi, which I do quite a lot. Applying > the filter > then subscribing to all events seemed the most stable and allowed me > to > subscribe to additional events should I need them. > > In the latest snapshot, I don't receive any events through this > mechanism. I > tried "event text all" as newly suggested on the ESL outbound wiki > page, but > this transmits a heap of white space to my socket then kills > freeswitch! > > Could somebody please tell me if this is something that is "work in > progress" and will ultimately resume old behaviour, or should I be > doing > something else to monitor events for my uuid and anything I spawn in > that > session? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 -- 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 !DSPAM:4c12710232935550910691! From mail at jankubr.com Fri Jun 11 13:54:42 2010 From: mail at jankubr.com (Jan Kubr) Date: Fri, 11 Jun 2010 22:54:42 +0200 Subject: [Freeswitch-users] DETECTED_SPEECH events while playing a file In-Reply-To: References: Message-ID: async: On Fri, Jun 11, 2010 at 7:18 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > are you using sync or async mode on the socket. > > > On Thu, Jun 10, 2010 at 5:07 PM, Jan Kubr wrote: > >> Hi, >> I'm trying to detect speech while playing a file over the event socket. >> >> I start detecting speech: >> >> call-command: execute >> execute-app-name: detect_speech >> execute-app-arg: pocketsphinx ivr_grammar ivr >> event-lock:true >> >> Then playback the file: >> >> call-command: execute >> execute-app-name: playback >> execute-app-arg: ivr.wav >> event-lock:false >> >> Then resume the speech detection: >> >> call-command: execute >> execute-app-name: detect_speech >> execute-app-arg: resume >> event-lock:true >> >> >> Now when I say something, I do see this in the FreeSWITCH console: >> >> 2010-06-10 23:29:06.994346 [DEBUG] mod_pocketsphinx.c:383 Recognized: ONE, >> Confidence: 100 >> >> But I don't receive the DETECTED_SPEECH events over the socket. >> >> However, if I wait until the file is played, that is until I see this in >> the FreeSWITCH console: >> >> 2010-06-10 20:48:50.512148 [DEBUG] switch_ivr_play_say.c:1428 done playing >> file >> >> and say something after that, I do get the DETECTED_SPEECH events. >> >> >> Is there a way I could receive these events while the file is being >> played? >> >> Thanks, >> Jan Kubr >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/a6f76735/attachment-0001.html From dswardstrom at remotelink.com Fri Jun 11 13:57:47 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Fri, 11 Jun 2010 15:57:47 -0500 (CDT) Subject: [Freeswitch-users] JavaScript ApiExecute return values In-Reply-To: <1240906883.198.1276289799923.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <343390804.200.1276289867952.JavaMail.root@srvr12.remotelinkml.com> One of the problems that I have with the Wiki is that some of the APIs return a value. In fact, it is possible that many return a Success/Failure if nothing else. However, the Wiki does not specify if any data is returned. The problem sometimes is not with the specified API as listed in the JavaScript Wiki page itself but the APIs that can be invoked from ApiExecute() or with session.execute(). For example, currently I need to use the url_encode() capability. I can find it in mod_commands: url_encode Url encode a string. Usage: url_encode Normally I would expect that the code would return the encoded string as a return parameter but this is not stated in the description. I.E., the Wiki entry should be: url_encode Url encode a string. Usage: encoded_string = url_encode It is possible that the implementation encodes the string and returns it in the same parameter (i.e., in place). I tried looking at the code, but still am too new at looking at the FreeSwitch code. Of course, I would try it/test it and will do this. But it would be useful if the Wiki was clearer. Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom From kward at binarysignal.com Fri Jun 11 14:03:37 2010 From: kward at binarysignal.com (Kurt Ward) Date: Fri, 11 Jun 2010 14:03:37 -0700 Subject: [Freeswitch-users] Change in behaviour for ESL events? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C567CA60751@cooper> References: <1275896373992-5147891.post@n2.nabble.com> <012001cb0662$cdbae390$6930aab0$@com> <00a801cb094d$6049eb60$20ddc220$@com> <331434B6-59EF-46D5-9A96-711D0F55CF2D@binarysignal.com>, <549CFEF87AEDE841A38E9D15EAB4C04C567CA60751@cooper> Message-ID: <23461C17-391C-4BAB-B480-CB23BC802DEB@binarysignal.com> I'm assuming you mean Git core.autocrlf=false, which I have tried with no success. I will put a new VM together with a clean install of everything and try again. On Jun 11, 2010, at 1:34 PM, Peter Olsson wrote: > I build in Windows almost every day - it works just fine. It's > probably the old "CRFL-trick" that's ithe cause of the problems... > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org > ] för Anthony Minessale [anthony.minessale at gmail.com] > Skickat: den 11 juni 2010 19:15 > Till: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] Change in behaviour for ESL events? > > If that is the case then you should have reported that.... > Most likely you missed some of the instructions for building for > windows with git because nobody else has reported it. > > > On Fri, Jun 11, 2010 at 12:06 PM, Kurt Ward >> wrote: > The Windows build is broken on Git. The last rev I was able to build > in VS 2008 Express, Win32 was SVN. > > On Jun 11, 2010, at 9:34 AM, Anthony Minessale wrote: > >> anybody using svn revisions should update. From anthony.minessale at gmail.com Fri Jun 11 14:07:46 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Jun 2010 16:07:46 -0500 Subject: [Freeswitch-users] DETECTED_SPEECH events while playing a file In-Reply-To: References: Message-ID: instead of waiting for the events to be fired, instead try sending at the beginning of your session divert_events\n\n that should cause the actual events to be queued to your channel in real time On Fri, Jun 11, 2010 at 3:54 PM, Jan Kubr wrote: > async: > > > > > On Fri, Jun 11, 2010 at 7:18 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> are you using sync or async mode on the socket. >> >> >> On Thu, Jun 10, 2010 at 5:07 PM, Jan Kubr wrote: >> >>> Hi, >>> I'm trying to detect speech while playing a file over the event socket. >>> >>> I start detecting speech: >>> >>> call-command: execute >>> execute-app-name: detect_speech >>> execute-app-arg: pocketsphinx ivr_grammar ivr >>> event-lock:true >>> >>> Then playback the file: >>> >>> call-command: execute >>> execute-app-name: playback >>> execute-app-arg: ivr.wav >>> event-lock:false >>> >>> Then resume the speech detection: >>> >>> call-command: execute >>> execute-app-name: detect_speech >>> execute-app-arg: resume >>> event-lock:true >>> >>> >>> Now when I say something, I do see this in the FreeSWITCH console: >>> >>> 2010-06-10 23:29:06.994346 [DEBUG] mod_pocketsphinx.c:383 Recognized: >>> ONE, Confidence: 100 >>> >>> But I don't receive the DETECTED_SPEECH events over the socket. >>> >>> However, if I wait until the file is played, that is until I see this in >>> the FreeSWITCH console: >>> >>> 2010-06-10 20:48:50.512148 [DEBUG] switch_ivr_play_say.c:1428 done >>> playing file >>> >>> and say something after that, I do get the DETECTED_SPEECH events. >>> >>> >>> Is there a way I could receive these events while the file is being >>> played? >>> >>> Thanks, >>> Jan Kubr >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100611/e8afc3dd/attachment.html From anthony.minessale at gmail.com Fri Jun 11 14:18:03 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Jun 2010 16:18:03 -0500 Subject: [Freeswitch-users] JavaScript ApiExecute return values In-Reply-To: <343390804.200.1276289867952.JavaMail.root@srvr12.remotelinkml.com> References: <1240906883.198.1276289799923.JavaMail.root@srvr12.remotelinkml.com> <343390804.200.1276289867952.JavaMail.root@srvr12.remotelinkml.com> Message-ID: all FSAPI interface calls are prototyped the same string in string out an easy way to try them is to type it at the cli url_encode 'this is a test' On Fri, Jun 11, 2010 at 3:57 PM, David Swardstrom < dswardstrom at remotelink.com> wrote: > One of the problems that I have with the Wiki is that some of the APIs > return a value. > In fact, it is possible that many return a Success/Failure if nothing else. > However, the Wiki does not specify if any data is returned. > > The problem sometimes is not with the specified API as listed in the > JavaScript Wiki page itself but the APIs that can be invoked from > ApiExecute() or with session.execute(). > > For example, currently I need to use the url_encode() capability. > I can find it in mod_commands: > > url_encode > Url encode a string. > Usage: url_encode > > > Normally I would expect that the code would return the encoded string as a > return > parameter but this is not stated in the description. I.E., the Wiki entry > should be: > > url_encode > Url encode a string. > Usage: encoded_string = url_encode > > It is possible that the implementation encodes the string and returns it > in the same parameter (i.e., in place). > > I tried looking at the code, but still am too new at looking at the > FreeSwitch code. > Of course, I would try it/test it and will do this. But it would be useful > if > the Wiki was clearer. > > Regards, > David Swardstrom > (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100611/9157e115/attachment.html From bwibowo at gmail.com Fri Jun 11 15:02:43 2010 From: bwibowo at gmail.com (budi wibowo) Date: Sat, 12 Jun 2010 05:02:43 +0700 Subject: [Freeswitch-users] calling between extension Message-ID: hi i have question regarding profile user in FS. in /usr/local/freeswitch/conf/directory/sip1.domain.com i have budi.xml containing and i have other profile for other user say james.xml. what need to be configured from FS so i can can call james directly from my softphone regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100612/307bfa71/attachment-0001.html From no-reply at dropbox.com Fri Jun 11 15:58:27 2010 From: no-reply at dropbox.com (Dropbox) Date: Fri, 11 Jun 2010 22:58:27 +0000 Subject: [Freeswitch-users] Andres Martin has invited you to Dropbox Message-ID: <20100611225827.1C6DB306EFA@mailman.dropbox.com> We're excited to let you know that Andres Martin has invited you to Dropbox! Andres Martin has been using Dropbox to sync and share files online and across computers, and thought you might want it too. Visit http://www.dropbox.com/link/20.9kABIoVFR8/NjIxODA4MDYyNw to get started. - The Dropbox Team ____________________________________________________ To stop receiving invites from Dropbox, please go to http://www.dropbox.com/bl/070b00e90e9d/freeswitch-users%40lists.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/0a4313de/attachment.html From helmut.kuper at ewetel.de Fri Jun 11 17:12:39 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Sat, 12 Jun 2010 02:12:39 +0200 Subject: [Freeswitch-users] ESL, phpmod Message-ID: <4C12D0F7.5000300@ewetel.de> Hello, I try to setup a php daemon which uses ESL. I run the sample php scripts successfully (inbound). Now I want to have it outbound to my php daemon socket. The forked child process which has to interact with incoming tcp connection from FS is started successfully. So in this state I have the client socket which I have to pass now somehow to ESLconnection I guess. FS ruby wiki gives this as an example: @con = ESL::ESLconnection.new(client_socket.fileno) In php I try this: $con = new ESLconnection($csock); As a result I got this error: PHP Fatal error: No matching function for overloaded 'new_ESLconnection' in /usr/share/pear/ESL.php on line 117 Any ideas? regards Helmut From helmut.kuper at ewetel.de Fri Jun 11 18:31:51 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Sat, 12 Jun 2010 03:31:51 +0200 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: <4C12D0F7.5000300@ewetel.de> References: <4C12D0F7.5000300@ewetel.de> Message-ID: <4C12E387.3030908@ewetel.de> Hello, when I type cast the socket var to integer and then do strace my php script I got this when new ESL connection is created: setsockopt(6, SOL_TCP, TCP_NODELAY, [1], 4) = -1 EBADF (Bad file descriptor) sendto(6, "connect\n\n", 9, 0, NULL, 0) = -1 EBADF (Bad file descriptor) There is no fatal error but the fd is somehow bad ... So it seems to me the parameter for "new ESLconnection" must be a number - but which? I tested both, sockets created with socket_create as well as with fsockopen. On 12.06.2010 02:12, Helmut Kuper wrote: > Hello, > > > I try to setup a php daemon which uses ESL. > > I run the sample php scripts successfully (inbound). Now I want to have > it outbound to my php daemon socket. > > The forked child process which has to interact with incoming tcp > connection from FS is started successfully. So in this state I have the > client socket which I have to pass now somehow to ESLconnection I guess. > > FS ruby wiki gives this as an example: > > @con = ESL::ESLconnection.new(client_socket.fileno) > > > In php I try this: > > $con = new ESLconnection($csock); > > As a result I got this error: > > PHP Fatal error: No matching function for overloaded > 'new_ESLconnection' in /usr/share/pear/ESL.php on line 117 > > > Any ideas? > > regards > Helmut > From nagalenoj at gmail.com Fri Jun 11 21:32:17 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Sat, 12 Jun 2010 10:02:17 +0530 Subject: [Freeswitch-users] leg_timeout isn't working. In-Reply-To: References: Message-ID: Anthony, Sorry for the miscommunication. Don't mistake me. I've posted in mailing lists because if some one else have done small workarounds and get this already solved, then I could get this solved easily. On Fri, Jun 11, 2010 at 9:55 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > please expect more than a 2 day turnaround using free support on a free > project. > I have, in fact fixed your problem which was specific to using the user/ > channel. > please have more patience in the future. Once you open a jira someone will > get to it eventually. > > > > On Fri, Jun 11, 2010 at 8:09 AM, Nagalenoj H. wrote: > >> Dear friends, >> I've posted an issue in jira(2 days back) but I didn't get any response >> there. I just want to confirm whether any one else is facing the same and it >> is really an issue. >> >> Description: >> When I tried to execute 'bridge [leg_timeout=10]user/1010', it doesn't >> quit ringing if the callee didn't respond in 10 seconds. But when I use it >> in {leg_timeout=10}user/1010, it's working. >> When I refer the wiki, it is given as it shouldn't be used in curly >> braces. >> >> 'Can be used in per-leg [], but not in global {} for the dialstring.' >> -- From wiki ( >> http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout) >> >> Attached the log here, >> http://jira.freeswitch.org/browse/MODAPP-433 >> >> -- >> Regards, >> Nagalenoj H. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 & Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100612/ff2205ad/attachment.html From babak.freeswitch at gmail.com Fri Jun 11 21:32:56 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Sat, 12 Jun 2010 09:02:56 +0430 Subject: [Freeswitch-users] help about a system with login logout Message-ID: Hi I need to implement a system in which users should login and logout to be able to place or receive calls using cisco 7941 ip phones on freeswitch (first I tried to somehow program one of soft keys as login and logout shortcut but I couldn't do that). Any suggestions would be appreciated. thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100612/10527c03/attachment.html From msc at freeswitch.org Fri Jun 11 22:06:40 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Jun 2010 22:06:40 -0700 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: <4C12D0F7.5000300@ewetel.de> References: <4C12D0F7.5000300@ewetel.de> Message-ID: Make sure the ESL is properly built and that the PHP mod is built and installed. If you did a "make current" recently the you'll need to rebuild your ESL stuff. -MC On Fri, Jun 11, 2010 at 5:12 PM, Helmut Kuper wrote: > Hello, > > > I try to setup a php daemon which uses ESL. > > I run the sample php scripts successfully (inbound). Now I want to have > it outbound to my php daemon socket. > > The forked child process which has to interact with incoming tcp > connection from FS is started successfully. So in this state I have the > client socket which I have to pass now somehow to ESLconnection I guess. > > FS ruby wiki gives this as an example: > > @con = ESL::ESLconnection.new(client_socket.fileno) > > > In php I try this: > > $con = new ESLconnection($csock); > > As a result I got this error: > > PHP Fatal error: No matching function for overloaded > 'new_ESLconnection' in /usr/share/pear/ESL.php on line 117 > > > Any ideas? > > regards > Helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100611/5ea09626/attachment.html From christian.loeschenkohl at xpirio.com Sat Jun 12 01:34:36 2010 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Sat, 12 Jun 2010 10:34:36 +0200 Subject: [Freeswitch-users] major problem with latest version Message-ID: <4C13469C.5040103@xpirio.com> hello i expirience a big problem with the latest version (pulled from git). inbound sip calls don't get answered (i do answer them with the answer app). does anybody expirience the same issue? br -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From mail at jankubr.com Sat Jun 12 02:54:38 2010 From: mail at jankubr.com (Jan Kubr) Date: Sat, 12 Jun 2010 11:54:38 +0200 Subject: [Freeswitch-users] DETECTED_SPEECH events while playing a file In-Reply-To: References: Message-ID: That did the trick, thanks a lot. Is there any downside to having divert_events on all the time? I've updated the wiki for others who might run into this issue: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_detect_speech Jan On Fri, Jun 11, 2010 at 11:07 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > instead of waiting for the events to be fired, instead try sending at the > beginning of your session > > divert_events\n\n > > that should cause the actual events to be queued to your channel in real > time > > > > > On Fri, Jun 11, 2010 at 3:54 PM, Jan Kubr wrote: > >> async: >> >> >> >> >> On Fri, Jun 11, 2010 at 7:18 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> are you using sync or async mode on the socket. >>> >>> >>> On Thu, Jun 10, 2010 at 5:07 PM, Jan Kubr wrote: >>> >>>> Hi, >>>> I'm trying to detect speech while playing a file over the event socket. >>>> >>>> I start detecting speech: >>>> >>>> call-command: execute >>>> execute-app-name: detect_speech >>>> execute-app-arg: pocketsphinx ivr_grammar ivr >>>> event-lock:true >>>> >>>> Then playback the file: >>>> >>>> call-command: execute >>>> execute-app-name: playback >>>> execute-app-arg: ivr.wav >>>> event-lock:false >>>> >>>> Then resume the speech detection: >>>> >>>> call-command: execute >>>> execute-app-name: detect_speech >>>> execute-app-arg: resume >>>> event-lock:true >>>> >>>> >>>> Now when I say something, I do see this in the FreeSWITCH console: >>>> >>>> 2010-06-10 23:29:06.994346 [DEBUG] mod_pocketsphinx.c:383 Recognized: >>>> ONE, Confidence: 100 >>>> >>>> But I don't receive the DETECTED_SPEECH events over the socket. >>>> >>>> However, if I wait until the file is played, that is until I see this in >>>> the FreeSWITCH console: >>>> >>>> 2010-06-10 20:48:50.512148 [DEBUG] switch_ivr_play_say.c:1428 done >>>> playing file >>>> >>>> and say something after that, I do get the DETECTED_SPEECH events. >>>> >>>> >>>> Is there a way I could receive these events while the file is being >>>> played? >>>> >>>> Thanks, >>>> Jan Kubr >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > > -- > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100612/c32a765b/attachment.html From msc at freeswitch.org Sat Jun 12 04:37:00 2010 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 12 Jun 2010 04:37:00 -0700 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: <4C13469C.5040103@xpirio.com> References: <4C13469C.5040103@xpirio.com> Message-ID: <1A6B403C-12A3-4050-9A71-EB4B2BDE1371@freeswitch.org> What happens when calls come in? Post a console log. -MC Sent from my iPhone On Jun 12, 2010, at 1:34 AM, Christian L?schenkohl wrote: > hello > > i expirience a big problem with the latest version (pulled from git). > inbound sip calls don't get answered (i do answer them with the > answer app). > does anybody expirience the same issue? > > br > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 helmut.kuper at ewetel.de Sat Jun 12 10:59:08 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Sat, 12 Jun 2010 19:59:08 +0200 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: References: <4C12D0F7.5000300@ewetel.de> Message-ID: <4C13CAEC.3020005@ewetel.de> Hi Michael, ESL is built and installed, phpmod is built and installed. inbound works, test scripts works It is just the problem of passing an existing socket to new ESLconnection. A socket resource is denied, an integer is accepted but is rejected with "bad file descriptor" Am 12.06.2010 07:06, schrieb Michael Collins: > Make sure the ESL is properly built and that the PHP mod is built and > installed. If you did a "make current" recently the you'll need to > rebuild your ESL stuff. > -MC > > On Fri, Jun 11, 2010 at 5:12 PM, Helmut Kuper > wrote: > > Hello, > > > I try to setup a php daemon which uses ESL. > > I run the sample php scripts successfully (inbound). Now I want to > have > it outbound to my php daemon socket. > > The forked child process which has to interact with incoming tcp > connection from FS is started successfully. So in this state I > have the > client socket which I have to pass now somehow to ESLconnection I > guess. > > FS ruby wiki gives this as an example: > > @con = ESL::ESLconnection.new(client_socket.fileno) > > > In php I try this: > > $con = new ESLconnection($csock); > > As a result I got this error: > > PHP Fatal error: No matching function for overloaded > 'new_ESLconnection' in /usr/share/pear/ESL.php on line 117 > > > Any ideas? > > regards > Helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100612/87101b52/attachment.html From anthony.minessale at gmail.com Sat Jun 12 11:39:34 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 12 Jun 2010 13:39:34 -0500 Subject: [Freeswitch-users] DETECTED_SPEECH events while playing a file In-Reply-To: References: Message-ID: no, that is what the command was intended to be used for. On Jun 12, 2010 5:00 AM, "Jan Kubr" wrote: That did the trick, thanks a lot. Is there any downside to having divert_events on all the time? I've updated the wiki for others who might run into this issue: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_detect_speech Jan On Fri, Jun 11, 2010 at 11:07 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > > inste... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100612/6e806b64/attachment-0001.html From tayeb.meftah at gmail.com Sat Jun 12 12:52:33 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sat, 12 Jun 2010 19:52:33 +0000 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: References: Message-ID: <4C13E581.2050507@gmail.com> hi, you can just use a pin code;) otherwise use the db api to modify a sqlite DB and login/logout users thanks babak yakhchali a ?crit : > Hi > I need to implement a system in which users should login and logout to > be able to place or receive calls using cisco 7941 ip phones on > freeswitch (first I tried to somehow program one of soft keys as login > and logout shortcut but I couldn't do that). Any suggestions would be > appreciated. > thanx > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 5192 (20100612) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 5192 (20100612) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From david.ponzone at gmail.com Sat Jun 12 13:51:47 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Sat, 12 Jun 2010 22:51:47 +0200 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: References: Message-ID: Babak, I would suggest implementing 2 specific extensions in the defaullt dialplan, like *55 and *66 (or anything you want of course). When the phone dials *55, it will send the call to a script asking for a numeric password as DTMF. You would then authenticate the userid/password pair (or callerid/ password pair, or else) in a custom DB. If authenticated, you have then 2 solutions to allow outgoing calls: -you have a default extension at the end of your dialplan, using a script to check if the caller is authenticated, and if so, bridging the call to the dialed number or -in the internal SQlite DB used by FS, you change the user_context used by the phone from default XML to allowed XML (or else). In allowed XML, there would be a default extension for outgoing calls, although in default XML, there would only be *55. There should be *66 in allowed XML also, to logout the user (reverting its user_context to default XML). The last option could be wrong, I am not sure if the user_context is available for dynamic alteration in the DB. And I am pretty sure there are other quite smarter ways. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 12/06/2010 ? 06:32, babak yakhchali a ?crit : > Hi > I need to implement a system in which users should login and logout > to be able to place or receive calls using cisco 7941 ip phones on > freeswitch (first I tried to somehow program one of soft keys as > login and logout shortcut but I couldn't do that). Any suggestions > would be appreciated. > thanx > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100612/0b80c142/attachment.html From babak.freeswitch at gmail.com Sat Jun 12 21:26:14 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Sun, 13 Jun 2010 08:56:14 +0430 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: References: Message-ID: Thanx all is there anyway to register or unregister a phone from freeswitch? or if anyone is familiar with cisco 7941, is there anyway like xmlObjects to force an ipphone to register or unregister? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/7bce2ad3/attachment.html From babak.freeswitch at gmail.com Sat Jun 12 22:57:18 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Sun, 13 Jun 2010 10:27:18 +0430 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: References: Message-ID: I need to map user login logout to sip registrations and unregistrations -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/ca43a408/attachment.html From david.ponzone at gmail.com Sun Jun 13 00:01:33 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Sun, 13 Jun 2010 09:01:33 +0200 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: References: Message-ID: <34D6A82A-5F15-48AF-BCA5-7201AD9CDFCE@gmail.com> I don't think it's a valid idea as you probably can't prevent a phone to register automatically at startup. So a user would just have to reboot a phone to get it registered and able to call. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 13/06/2010 ? 07:57, babak yakhchali a ?crit : > I need to map user login logout to sip registrations and > unregistrations _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/c3a030ca/attachment-0001.html From christian.loeschenkohl at xpirio.com Sun Jun 13 00:40:45 2010 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Sun, 13 Jun 2010 09:40:45 +0200 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: <4C13469C.5040103@xpirio.com> References: <4C13469C.5040103@xpirio.com> Message-ID: <4C148B7D.6020208@xpirio.com> hello the problem is the following fs sends in actual git head P-Asserted-Identity: "43720570500" . the git version before (around 7 june) did send (with the same config/server) P-Asserted-Identity: "43720570500" . so the call does not get answered pastebin of the two 200 ok packets - http://pastebin.freeswitch.org/13175 fyi: the system at 93.185.139.77 is a sonus switch br On 2010-06-12 10:34, Christian L?schenkohl wrote: > hello > > i expirience a big problem with the latest version (pulled from git). > inbound sip calls don't get answered (i do answer them with the answer app). > does anybody expirience the same issue? > > br > -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From babak.freeswitch at gmail.com Sun Jun 13 02:10:44 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Sun, 13 Jun 2010 13:40:44 +0430 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: <34D6A82A-5F15-48AF-BCA5-7201AD9CDFCE@gmail.com> References: <34D6A82A-5F15-48AF-BCA5-7201AD9CDFCE@gmail.com> Message-ID: Now there is another problem! How can I change the user context? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/ff13c6d2/attachment.html From kdjakovic at hotmail.com Sun Jun 13 03:40:15 2010 From: kdjakovic at hotmail.com (katarina djakovic) Date: Sun, 13 Jun 2010 03:40:15 -0700 Subject: [Freeswitch-users] Missing CDRs with ATTENDED TRANSFER In-Reply-To: References: , <024a01cb0993$1194d480$34be7d80$@com>, Message-ID: Thanks guys, that solved my problem. All the best, Katarina Date: Fri, 11 Jun 2010 12:34:27 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Missing CDRs with ATTENDED TRANSFER On Fri, Jun 11, 2010 at 11:22 AM, Peder wrote: What value do you have for ?legs? in cdr_csv.conf.xml? This is just a guess, but you probably need to have AB to get the third cdr. Just be warned that you probably will end up with *four* CDRs if you turn on the b-leg logging. I've tested these scenarios with Polycoms and Snoms and it is pretty consistent. Attended transfers are a challenge for CDRs but once you see the b-legs' CDRs you'll start to grasp it. As an exercise, turn on the b-leg logging and then do some experiments. For instance, watch your CDRs at these points during the call flow: After A dials B and B answers but *before* A actually transfers the call to B After A transfer the call to B but before B and C disconnect After B and C disconnect When you see four CDRs for what is essentially a single phone call it helps to know at what stage in the call flow that each of the records was created. Have fun. :) -MC _________________________________________________________________ Hotmail is redefining busy with tools for the New Busy. Get more from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/e5a2404f/attachment.html From sos at sokhapkin.dyndns.org Sun Jun 13 08:42:55 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 13 Jun 2010 11:42:55 -0400 Subject: [Freeswitch-users] How to limit call attemp time? Message-ID: <201006131142.55400.sos@sokhapkin.dyndns.org> Which variable should be set to exit bridge application if the call has not been answered within the specified time? call_timeout variable works if called SIP end point responds with "180 Ringing", but doesn't work if the endpoint responds with early media. From codeghar at gmail.com Sun Jun 13 08:56:35 2010 From: codeghar at gmail.com (Code Ghar) Date: Sun, 13 Jun 2010 10:56:35 -0500 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: <201006131142.55400.sos@sokhapkin.dyndns.org> References: <201006131142.55400.sos@sokhapkin.dyndns.org> Message-ID: Hi Sergey Have you tried ignore_early_media? It may help. You can find more information from http://wiki.freeswitch.org/wiki/Channel_Variables On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin wrote: > Which variable should be set to exit bridge application if the call has not > been answered within the specified time? call_timeout variable works if > called > SIP end point responds with "180 Ringing", but doesn't work if the endpoint > responds with early media. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/3d3b6f55/attachment.html From sos at sokhapkin.dyndns.org Sun Jun 13 09:06:02 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 13 Jun 2010 12:06:02 -0400 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: References: <201006131142.55400.sos@sokhapkin.dyndns.org> Message-ID: <201006131206.02442.sos@sokhapkin.dyndns.org> Yes, I tried ignore_early_media. call_timeout doesn't work if early media received. On Sunday 13 June 2010, Code Ghar wrote: > Hi Sergey > > Have you tried ignore_early_media? It may help. You can find more > information from http://wiki.freeswitch.org/wiki/Channel_Variables > > On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > > wrote: > > Which variable should be set to exit bridge application if the call has > > not been answered within the specified time? call_timeout variable works > > if called > > SIP end point responds with "180 Ringing", but doesn't work if the > > endpoint responds with early media. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 jmesquita at freeswitch.org Sun Jun 13 09:16:17 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 13 Jun 2010 13:16:17 -0300 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: <201006131206.02442.sos@sokhapkin.dyndns.org> References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131206.02442.sos@sokhapkin.dyndns.org> Message-ID: Look at leg_progress_timeout or just progress_timeout. On Sunday, June 13, 2010, Sergey Okhapkin wrote: > Yes, I tried ignore_early_media. call_timeout doesn't work if early media > received. > > On Sunday 13 June 2010, Code Ghar wrote: >> Hi Sergey >> >> Have you tried ignore_early_media? It may help. You can find more >> information from http://wiki.freeswitch.org/wiki/Channel_Variables >> >> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin >> >> wrote: >> > Which variable should be set to exit bridge application if the call has >> > not been answered within the specified time? call_timeout variable works >> > if called >> > SIP end point responds with "180 Ringing", but doesn't work if the >> > endpoint responds with early media. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.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 > -- Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 From sos at sokhapkin.dyndns.org Sun Jun 13 09:25:32 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 13 Jun 2010 12:25:32 -0400 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131206.02442.sos@sokhapkin.dyndns.org> Message-ID: <201006131225.32231.sos@sokhapkin.dyndns.org> progress_timeout exits the bridge if NO 180 or 183 received, I'm receiving 183. On Sunday 13 June 2010, Jo?o Mesquita wrote: > Look at leg_progress_timeout or just progress_timeout. > > On Sunday, June 13, 2010, Sergey Okhapkin wrote: > > Yes, I tried ignore_early_media. call_timeout doesn't work if early media > > received. > > > > On Sunday 13 June 2010, Code Ghar wrote: > >> Hi Sergey > >> > >> Have you tried ignore_early_media? It may help. You can find more > >> information from http://wiki.freeswitch.org/wiki/Channel_Variables > >> > >> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > >> > >> wrote: > >> > Which variable should be set to exit bridge application if the call > >> > has not been answered within the specified time? call_timeout variable > >> > works if called > >> > SIP end point responds with "180 Ringing", but doesn't work if the > >> > endpoint responds with early media. > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users 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 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 david.ponzone at gmail.com Sun Jun 13 10:53:38 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Sun, 13 Jun 2010 19:53:38 +0200 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: References: <34D6A82A-5F15-48AF-BCA5-7201AD9CDFCE@gmail.com> Message-ID: <2D0975D5-188A-4CFC-A4DF-47AE82C71C41@gmail.com> You should start by accessing FS's internal SQlite DB and check what is there. It could be documented somewhere on the wiki, though. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 13/06/2010 ? 11:10, babak yakhchali a ?crit : > Now there is another problem! > How can I change the user context? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/92c4498f/attachment-0001.html From david.ponzone at gmail.com Sun Jun 13 10:56:34 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Sun, 13 Jun 2010 19:56:34 +0200 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: <201006131225.32231.sos@sokhapkin.dyndns.org> References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131206.02442.sos@sokhapkin.dyndns.org> <201006131225.32231.sos@sokhapkin.dyndns.org> Message-ID: What you want is a way to limit the ringing time isnt it ? I think it's not possible. I needed that some time ago, and I never found out the solution. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 13/06/2010 ? 18:25, Sergey Okhapkin a ?crit : > progress_timeout exits the bridge if NO 180 or 183 received, I'm > receiving > 183. > > On Sunday 13 June 2010, Jo?o Mesquita wrote: >> Look at leg_progress_timeout or just progress_timeout. >> >> On Sunday, June 13, 2010, Sergey Okhapkin >> wrote: >>> Yes, I tried ignore_early_media. call_timeout doesn't work if >>> early media >>> received. >>> >>> On Sunday 13 June 2010, Code Ghar wrote: >>>> Hi Sergey >>>> >>>> Have you tried ignore_early_media? It may help. You can find more >>>> information from http://wiki.freeswitch.org/wiki/Channel_Variables >>>> >>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin >>>> >>>> wrote: >>>>> Which variable should be set to exit bridge application if the >>>>> call >>>>> has not been answered within the specified time? call_timeout >>>>> variable >>>>> works if called >>>>> SIP end point responds with "180 Ringing", but doesn't work if the >>>>> endpoint responds with early media. >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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/20100613/cb405cf2/attachment.html From oseslija at gmail.com Sun Jun 13 10:59:06 2010 From: oseslija at gmail.com (Ognjen Seslija) Date: Sun, 13 Jun 2010 19:59:06 +0200 Subject: [Freeswitch-users] Missing CDRs with ATTENDED TRANSFER In-Reply-To: References: <024a01cb0993$1194d480$34be7d80$@com> Message-ID: Zdravo, Moras ukljuciti logovanje b-lega. Pozdrav, Ognjen On Jun 13, 2010 2:22 PM, "katarina djakovic" wrote: Thanks guys, that solved my problem. All the best, Katarina ------------------------------ Date: Fri, 11 Jun 2010 12:34:27 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Missing CDRs with ATTENDED TRANSFER On Fri, Jun 11, 2010 at 11:22 AM, Peder wrote: > > What value do you... ------------------------------ Hotmail is redefining busy with tools for the New Busy. Get more from your inbox. See how. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/aa59b6bc/attachment.html From sos at sokhapkin.dyndns.org Sun Jun 13 11:05:05 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 13 Jun 2010 14:05:05 -0400 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131225.32231.sos@sokhapkin.dyndns.org> Message-ID: <201006131405.05506.sos@sokhapkin.dyndns.org> Correct, I need to limit ringing time and continue dialplan execution if no answer within N seconds. It's trivial "find me" service, serial DID forwarding to multiple PSTN numbers. On Sunday 13 June 2010, David Ponzone wrote: > What you want is a way to limit the ringing time isnt it ? > I think it's not possible. I needed that some time ago, and I never > found out the solution. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > ? l'intention exclusive de ses destinataires. Toute utilisation ou > diffusion non autoris?e est interdite. Tout message ?lectronique est > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes > pas destinataire de ce message, merci de le d?truire imm?diatement et > d'avertir l'exp?diteur. > > Le 13/06/2010 ? 18:25, Sergey Okhapkin a ?crit : > > progress_timeout exits the bridge if NO 180 or 183 received, I'm > > receiving > > 183. > > > > On Sunday 13 June 2010, Jo?o Mesquita wrote: > >> Look at leg_progress_timeout or just progress_timeout. > >> > >> On Sunday, June 13, 2010, Sergey Okhapkin > >> > >> wrote: > >>> Yes, I tried ignore_early_media. call_timeout doesn't work if > >>> early media > >>> received. > >>> > >>> On Sunday 13 June 2010, Code Ghar wrote: > >>>> Hi Sergey > >>>> > >>>> Have you tried ignore_early_media? It may help. You can find more > >>>> information from http://wiki.freeswitch.org/wiki/Channel_Variables > >>>> > >>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > >>>> > >>>> wrote: > >>>>> Which variable should be set to exit bridge application if the > >>>>> call > >>>>> has not been answered within the specified time? call_timeout > >>>>> variable > >>>>> works if called > >>>>> SIP end point responds with "180 Ringing", but doesn't work if the > >>>>> endpoint responds with early media. > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > >>>>>e rs http://www.freeswitch.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-user > >>>s http://www.freeswitch.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 > From infos at madovsky.org Sun Jun 13 11:15:24 2010 From: infos at madovsky.org (Madovsky) Date: Sun, 13 Jun 2010 14:15:24 -0400 Subject: [Freeswitch-users] How to limit call attemp time? References: <201006131142.55400.sos@sokhapkin.dyndns.org><201006131225.32231.sos@sokhapkin.dyndns.org> <201006131405.05506.sos@sokhapkin.dyndns.org> Message-ID: <2E0E8DC7F9894A958DC618B04EC5E3CD@MOBILEE1705> Sergey, I did this before bridge and works for me Franck ----- Original Message ----- From: "Sergey Okhapkin" To: Sent: Sunday, June 13, 2010 2:05 PM Subject: Re: [Freeswitch-users] How to limit call attemp time? Correct, I need to limit ringing time and continue dialplan execution if no answer within N seconds. It's trivial "find me" service, serial DID forwarding to multiple PSTN numbers. On Sunday 13 June 2010, David Ponzone wrote: > What you want is a way to limit the ringing time isnt it ? > I think it's not possible. I needed that some time ago, and I never > found out the solution. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > ? l'intention exclusive de ses destinataires. Toute utilisation ou > diffusion non autoris?e est interdite. Tout message ?lectronique est > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes > pas destinataire de ce message, merci de le d?truire imm?diatement et > d'avertir l'exp?diteur. > > Le 13/06/2010 ? 18:25, Sergey Okhapkin a ?crit : > > progress_timeout exits the bridge if NO 180 or 183 received, I'm > > receiving > > 183. > > > > On Sunday 13 June 2010, Jo?o Mesquita wrote: > >> Look at leg_progress_timeout or just progress_timeout. > >> > >> On Sunday, June 13, 2010, Sergey Okhapkin > >> > >> wrote: > >>> Yes, I tried ignore_early_media. call_timeout doesn't work if > >>> early media > >>> received. > >>> > >>> On Sunday 13 June 2010, Code Ghar wrote: > >>>> Hi Sergey > >>>> > >>>> Have you tried ignore_early_media? It may help. You can find more > >>>> information from http://wiki.freeswitch.org/wiki/Channel_Variables > >>>> > >>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > >>>> > >>>> wrote: > >>>>> Which variable should be set to exit bridge application if the > >>>>> call > >>>>> has not been answered within the specified time? call_timeout > >>>>> variable > >>>>> works if called > >>>>> SIP end point responds with "180 Ringing", but doesn't work if the > >>>>> endpoint responds with early media. > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > >>>>>e rs http://www.freeswitch.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-user > >>>s http://www.freeswitch.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 > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sos at sokhapkin.dyndns.org Sun Jun 13 11:24:51 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 13 Jun 2010 14:24:51 -0400 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: <2E0E8DC7F9894A958DC618B04EC5E3CD@MOBILEE1705> References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131405.05506.sos@sokhapkin.dyndns.org> <2E0E8DC7F9894A958DC618B04EC5E3CD@MOBILEE1705> Message-ID: <201006131424.51953.sos@sokhapkin.dyndns.org> Are you getting SIP 180 or SIP 183 from the B leg? Everything works fine when SIP 180 is received, but call_timeout doesn't work when SIP 183 is received. switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER condition, "media ready" includes early media also :-( On Sunday 13 June 2010, Madovsky wrote: > Sergey, > > I did this > > > > > > before bridge and works for me > > Franck > > ----- Original Message ----- > From: "Sergey Okhapkin" > To: > Sent: Sunday, June 13, 2010 2:05 PM > Subject: Re: [Freeswitch-users] How to limit call attemp time? > > > Correct, I need to limit ringing time and continue dialplan execution if no > answer within N seconds. > > It's trivial "find me" service, serial DID forwarding to multiple PSTN > numbers. > > On Sunday 13 June 2010, David Ponzone wrote: > > What you want is a way to limit the ringing time isnt it ? > > I think it's not possible. I needed that some time ago, and I never > > found out the solution. > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > > ? l'intention exclusive de ses destinataires. Toute utilisation ou > > diffusion non autoris?e est interdite. Tout message ?lectronique est > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes > > pas destinataire de ce message, merci de le d?truire imm?diatement et > > d'avertir l'exp?diteur. > > > > Le 13/06/2010 ? 18:25, Sergey Okhapkin a ?crit : > > > progress_timeout exits the bridge if NO 180 or 183 received, I'm > > > receiving > > > 183. > > > > > > On Sunday 13 June 2010, Jo?o Mesquita wrote: > > >> Look at leg_progress_timeout or just progress_timeout. > > >> > > >> On Sunday, June 13, 2010, Sergey Okhapkin > > >> > > >> wrote: > > >>> Yes, I tried ignore_early_media. call_timeout doesn't work if > > >>> early media > > >>> received. > > >>> > > >>> On Sunday 13 June 2010, Code Ghar wrote: > > >>>> Hi Sergey > > >>>> > > >>>> Have you tried ignore_early_media? It may help. You can find more > > >>>> information from http://wiki.freeswitch.org/wiki/Channel_Variables > > >>>> > > >>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > > >>>> > > >>>> wrote: > > >>>>> Which variable should be set to exit bridge application if the > > >>>>> call > > >>>>> has not been answered within the specified time? call_timeout > > >>>>> variable > > >>>>> works if called > > >>>>> SIP end point responds with "180 Ringing", but doesn't work if the > > >>>>> endpoint responds with early media. > > >>>>> > > >>>>> _______________________________________________ > > >>>>> FreeSWITCH-users mailing list > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > >>>>>us e rs http://www.freeswitch.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-us > > >>>er s http://www.freeswitch.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-user > > >s http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 peder at networkoblivion.com Sun Jun 13 11:43:47 2010 From: peder at networkoblivion.com (Peder) Date: Sun, 13 Jun 2010 13:43:47 -0500 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: <201006131424.51953.sos@sokhapkin.dyndns.org> References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131405.05506.sos@sokhapkin.dyndns.org> <2E0E8DC7F9894A958DC618B04EC5E3CD@MOBILEE1705> <201006131424.51953.sos@sokhapkin.dyndns.org> Message-ID: <057101cb0b28$5be87730$13b96590$@com> How about leg_timeout on that specific leg of the call. I had an issue where a general call_timeout didn't work if there was no response but leg_timeout did work. Not sure exactly how it works if there is a response though. Worth a shot. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Okhapkin Sent: Sunday, June 13, 2010 1:25 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How to limit call attemp time? Are you getting SIP 180 or SIP 183 from the B leg? Everything works fine when SIP 180 is received, but call_timeout doesn't work when SIP 183 is received. switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER condition, "media ready" includes early media also :-( On Sunday 13 June 2010, Madovsky wrote: > Sergey, > > I did this > > > repk > > > before bridge and works for me > > Franck > > ----- Original Message ----- > From: "Sergey Okhapkin" > To: > Sent: Sunday, June 13, 2010 2:05 PM > Subject: Re: [Freeswitch-users] How to limit call attemp time? > > > Correct, I need to limit ringing time and continue dialplan execution if no > answer within N seconds. > > It's trivial "find me" service, serial DID forwarding to multiple PSTN > numbers. > > On Sunday 13 June 2010, David Ponzone wrote: > > What you want is a way to limit the ringing time isnt it ? > > I think it's not possible. I needed that some time ago, and I never > > found out the solution. > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > > ? l'intention exclusive de ses destinataires. Toute utilisation ou > > diffusion non autoris?e est interdite. Tout message ?lectronique est > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes > > pas destinataire de ce message, merci de le d?truire imm?diatement et > > d'avertir l'exp?diteur. > > > > Le 13/06/2010 ? 18:25, Sergey Okhapkin a ?crit : > > > progress_timeout exits the bridge if NO 180 or 183 received, I'm > > > receiving > > > 183. > > > > > > On Sunday 13 June 2010, Jo?o Mesquita wrote: > > >> Look at leg_progress_timeout or just progress_timeout. > > >> > > >> On Sunday, June 13, 2010, Sergey Okhapkin > > >> > > >> wrote: > > >>> Yes, I tried ignore_early_media. call_timeout doesn't work if > > >>> early media > > >>> received. > > >>> > > >>> On Sunday 13 June 2010, Code Ghar wrote: > > >>>> Hi Sergey > > >>>> > > >>>> Have you tried ignore_early_media? It may help. You can find more > > >>>> information from http://wiki.freeswitch.org/wiki/Channel_Variables > > >>>> > > >>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > > >>>> > > >>>> wrote: > > >>>>> Which variable should be set to exit bridge application if the > > >>>>> call > > >>>>> has not been answered within the specified time? call_timeout > > >>>>> variable > > >>>>> works if called > > >>>>> SIP end point responds with "180 Ringing", but doesn't work if the > > >>>>> endpoint responds with early media. > > >>>>> > > >>>>> _______________________________________________ > > >>>>> FreeSWITCH-users mailing list > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > >>>>>us e rs http://www.freeswitch.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-us > > >>>er s http://www.freeswitch.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-user > > >s http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 From oseslija at gmail.com Sun Jun 13 11:46:15 2010 From: oseslija at gmail.com (Ognjen Seslija) Date: Sun, 13 Jun 2010 20:46:15 +0200 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: References: Message-ID: I don't think cisco does REGISTER with login/logout rather than SUBSCRIBE that needs to be challenged and authenticated. With current code I don't think it's possible. On Sun, Jun 13, 2010 at 6:26 AM, babak yakhchali wrote: > Thanx all > is there anyway to register or unregister a phone from freeswitch? or if > anyone is familiar with cisco 7941, is there anyway like xmlObjects to force > an ipphone to register or unregister? > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/b5a7b0e7/attachment.html From sos at sokhapkin.dyndns.org Sun Jun 13 11:56:38 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 13 Jun 2010 14:56:38 -0400 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: <057101cb0b28$5be87730$13b96590$@com> References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131424.51953.sos@sokhapkin.dyndns.org> <057101cb0b28$5be87730$13b96590$@com> Message-ID: <201006131456.38197.sos@sokhapkin.dyndns.org> Just tried, leg_timeout doesn't work too. On Sunday 13 June 2010, Peder wrote: > How about leg_timeout on that specific leg of the call. I had an issue > where a general call_timeout didn't work if there was no response but > leg_timeout did work. Not sure exactly how it works if there is a response > though. Worth a shot. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey > Okhapkin > Sent: Sunday, June 13, 2010 1:25 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] How to limit call attemp time? > > Are you getting SIP 180 or SIP 183 from the B leg? Everything works fine > when > SIP 180 is received, but call_timeout doesn't work when SIP 183 is > received. > > switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER > condition, "media ready" includes early media also :-( > > On Sunday 13 June 2010, Madovsky wrote: > > Sergey, > > > > I did this > > > > > > repk > > > > > > before bridge and works for me > > > > Franck > > > > ----- Original Message ----- > > From: "Sergey Okhapkin" > > To: > > Sent: Sunday, June 13, 2010 2:05 PM > > Subject: Re: [Freeswitch-users] How to limit call attemp time? > > > > > > Correct, I need to limit ringing time and continue dialplan execution if > > no > > > answer within N seconds. > > > > It's trivial "find me" service, serial DID forwarding to multiple PSTN > > numbers. > > > > On Sunday 13 June 2010, David Ponzone wrote: > > > What you want is a way to limit the ringing time isnt it ? > > > I think it's not possible. I needed that some time ago, and I never > > > found out the solution. > > > > > > David Ponzone Direction Technique > > > email: david.ponzone at ipeva.fr > > > tel: 01 74 03 18 97 > > > gsm: 06 66 98 76 34 > > > > > > Service Client IPeva > > > tel: 0811 46 26 26 > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > > > ? l'intention exclusive de ses destinataires. Toute utilisation ou > > > diffusion non autoris?e est interdite. Tout message ?lectronique est > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes > > > pas destinataire de ce message, merci de le d?truire imm?diatement et > > > d'avertir l'exp?diteur. > > > > > > Le 13/06/2010 ? 18:25, Sergey Okhapkin a ?crit : > > > > progress_timeout exits the bridge if NO 180 or 183 received, I'm > > > > receiving > > > > 183. > > > > > > > > On Sunday 13 June 2010, Jo?o Mesquita wrote: > > > >> Look at leg_progress_timeout or just progress_timeout. > > > >> > > > >> On Sunday, June 13, 2010, Sergey Okhapkin > > > >> > > > >> wrote: > > > >>> Yes, I tried ignore_early_media. call_timeout doesn't work if > > > >>> early media > > > >>> received. > > > >>> > > > >>> On Sunday 13 June 2010, Code Ghar wrote: > > > >>>> Hi Sergey > > > >>>> > > > >>>> Have you tried ignore_early_media? It may help. You can find more > > > >>>> information from http://wiki.freeswitch.org/wiki/Channel_Variables > > > >>>> > > > >>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > > > >>>> > > > >>>> wrote: > > > >>>>> Which variable should be set to exit bridge application if the > > > >>>>> call > > > >>>>> has not been answered within the specified time? call_timeout > > > >>>>> variable > > > >>>>> works if called > > > >>>>> SIP end point responds with "180 Ringing", but doesn't work if > > > >>>>> the endpoint responds with early media. > > > >>>>> > > > >>>>> _______________________________________________ > > > >>>>> FreeSWITCH-users mailing list > > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > > > >>>>>us e rs http://www.freeswitch.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-us > > > > >>>er s http://www.freeswitch.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-user > > > > >s http://www.freeswitch.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 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brokendash at gmail.com Sat Jun 12 04:58:42 2010 From: brokendash at gmail.com (broken dash) Date: Sat, 12 Jun 2010 06:58:42 -0500 Subject: [Freeswitch-users] Playback of Multiple Items Message-ID: Hello, Does there happen to be a way to configure FS to playback multiple shoutcast streams to an endpoint? I've tried making two extensions setup like so... My endpoints can dial each single ext without issue but the combine ext only seems to grab the first bridge xml entry and I've tried several variations of bridging multiple endpoints, etc.. Perhaps I'm approaching this incorrectly? Thanks, Brian M. == From errotan at elder.hu Sat Jun 12 05:39:49 2010 From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=) Date: Sat, 12 Jun 2010 14:39:49 +0200 Subject: [Freeswitch-users] calling between extension In-Reply-To: References: Message-ID: <201006121439.49226.errotan@elder.hu> 2010. j?nius 12. 00.02.43 d?tummal budi wibowo az al?bbiakat ?rta: > hi > i have question regarding profile user in FS. > in /usr/local/freeswitch/conf/directory/sip1.domain.com i have budi.xml > containing > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > and i have other profile for other user say james.xml. > what need to be configured from FS so i can can call james directly from my > softphone > > regards Hi. Please look at the wiki: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML You should need something like this: Create a new xml file dialplan/default/ directory and put these lines in it: The above calls "james" when you dial 1030. From errotan at elder.hu Sat Jun 12 07:18:28 2010 From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=) Date: Sat, 12 Jun 2010 16:18:28 +0200 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: References: Message-ID: <201006121618.28688.errotan@elder.hu> 2010. j?nius 12. 06.32.56 d?tummal babak yakhchali az al?bbiakat ?rta: > Hi > I need to implement a system in which users should login and logout to be > able to place or receive calls using cisco 7941 ip phones on freeswitch > (first I tried to somehow program one of soft keys as login and logout > shortcut but I couldn't do that). Any suggestions would be appreciated. > thanx Hi. You can write some Lua scripts ( http://wiki.freeswitch.org/wiki/Lua ) to autheticate users and store which extension they logged in. From brokendash at gmail.com Sat Jun 12 18:47:56 2010 From: brokendash at gmail.com (broken dash) Date: Sat, 12 Jun 2010 20:47:56 -0500 Subject: [Freeswitch-users] Playback of Multiple Items In-Reply-To: References: Message-ID: Any ideas on how to get this to work? I'm a noob so please let me know if I need to attach any additional XML config info... Thanks Brian M. On Sat, Jun 12, 2010 at 6:58 AM, broken dash wrote: > Hello, > > Does there happen to be a way to configure FS to playback multiple > shoutcast streams to an endpoint? I've tried making two extensions > setup like so... ?My endpoints can dial each single ext without issue > but the combine ext only seems to grab the first bridge xml entry and > I've tried several variations of bridging multiple endpoints, etc.. > Perhaps I'm approaching this incorrectly? > > Thanks, > Brian M. > > == > > > ? > ? ? ? > ? ? ? data="shout://relay.someserver.com/feed01"/> > ? > > > ? > ? ? ? > ? ? ? data="shout://relay.someserver.com/feed02"/> > > > ? > ? ? ? > ? ? ? > ? > From errotan at elder.hu Sun Jun 13 04:18:03 2010 From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=) Date: Sun, 13 Jun 2010 13:18:03 +0200 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: References: <34D6A82A-5F15-48AF-BCA5-7201AD9CDFCE@gmail.com> Message-ID: <201006131318.03752.errotan@elder.hu> 2010. j?nius 13. 11.10.44 d?tummal babak yakhchali az al?bbiakat ?rta: > Now there is another problem! > How can I change the user context? You can't change user context but you can use transfer: If you authenticated the user. From codeghar at gmail.com Sun Jun 13 14:26:02 2010 From: codeghar at gmail.com (Code Ghar) Date: Sun, 13 Jun 2010 16:26:02 -0500 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: <201006131456.38197.sos@sokhapkin.dyndns.org> References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131424.51953.sos@sokhapkin.dyndns.org> <057101cb0b28$5be87730$13b96590$@com> <201006131456.38197.sos@sokhapkin.dyndns.org> Message-ID: I tested it the following way: After 10 seconds, FS sends "cancel" to gateway1 and sends "invite" to gateway2 after sleeping for a while (optional). Once it sends invite to gateway2, after 15 seconds FS sends "cancel" to gateway2 and sends "invite" to gateway3. If after 30 seconds call is not answered it sends "cancel" to gateway3 and then does not call any other gateway. From all three gateways I received "183 Session Progress". If at any point call is answered, then subsequent gateways are not tried. For example, if call is answered through gateway1, then after hangup it doesn't attempt gateway2 and gateway3. Now comes your situation, where you need to try one gateway only. In this situation the following may be of help. As you can see, we have eliminated all gateways except gateway1. I have tested this myself and after 10 seconds FS sends "cancel" to gateway1 and call is disconnected. Hope this helps. I used the examples at following wiki links to create and test the above configurations. Misc. Dialplan Tools bridge http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge Dialplan FollowMe http://wiki.freeswitch.org/wiki/Dialplan_FollowMe On Sun, Jun 13, 2010 at 1:56 PM, Sergey Okhapkin wrote: > Just tried, leg_timeout doesn't work too. > > On Sunday 13 June 2010, Peder wrote: > > How about leg_timeout on that specific leg of the call. I had an issue > > where a general call_timeout didn't work if there was no response but > > leg_timeout did work. Not sure exactly how it works if there is a > response > > though. Worth a shot. > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Sergey > > Okhapkin > > Sent: Sunday, June 13, 2010 1:25 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] How to limit call attemp time? > > > > Are you getting SIP 180 or SIP 183 from the B leg? Everything works fine > > when > > SIP 180 is received, but call_timeout doesn't work when SIP 183 is > > received. > > > > switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER > > condition, "media ready" includes early media also :-( > > > > On Sunday 13 June 2010, Madovsky wrote: > > > Sergey, > > > > > > I did this > > > > > > > > > repk > > > > > > > > > before bridge and works for me > > > > > > Franck > > > > > > ----- Original Message ----- > > > From: "Sergey Okhapkin" > > > To: > > > Sent: Sunday, June 13, 2010 2:05 PM > > > Subject: Re: [Freeswitch-users] How to limit call attemp time? > > > > > > > > > Correct, I need to limit ringing time and continue dialplan execution > if > > > > no > > > > > answer within N seconds. > > > > > > It's trivial "find me" service, serial DID forwarding to multiple PSTN > > > numbers. > > > > > > On Sunday 13 June 2010, David Ponzone wrote: > > > > What you want is a way to limit the ringing time isnt it ? > > > > I think it's not possible. I needed that some time ago, and I never > > > > found out the solution. > > > > > > > > David Ponzone Direction Technique > > > > email: david.ponzone at ipeva.fr > > > > tel: 01 74 03 18 97 > > > > gsm: 06 66 98 76 34 > > > > > > > > Service Client IPeva > > > > tel: 0811 46 26 26 > > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > > > > ? l'intention exclusive de ses destinataires. Toute utilisation ou > > > > diffusion non autoris?e est interdite. Tout message ?lectronique est > > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre > > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes > > > > pas destinataire de ce message, merci de le d?truire imm?diatement et > > > > d'avertir l'exp?diteur. > > > > > > > > Le 13/06/2010 ? 18:25, Sergey Okhapkin a ?crit : > > > > > progress_timeout exits the bridge if NO 180 or 183 received, I'm > > > > > receiving > > > > > 183. > > > > > > > > > > On Sunday 13 June 2010, Jo?o Mesquita wrote: > > > > >> Look at leg_progress_timeout or just progress_timeout. > > > > >> > > > > >> On Sunday, June 13, 2010, Sergey Okhapkin > > > > >> > > > > >> wrote: > > > > >>> Yes, I tried ignore_early_media. call_timeout doesn't work if > > > > >>> early media > > > > >>> received. > > > > >>> > > > > >>> On Sunday 13 June 2010, Code Ghar wrote: > > > > >>>> Hi Sergey > > > > >>>> > > > > >>>> Have you tried ignore_early_media? It may help. You can find > more > > > > >>>> information from > http://wiki.freeswitch.org/wiki/Channel_Variables > > > > >>>> > > > > >>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > > > > >>>> > > > > >>>> wrote: > > > > >>>>> Which variable should be set to exit bridge application if the > > > > >>>>> call > > > > >>>>> has not been answered within the specified time? call_timeout > > > > >>>>> variable > > > > >>>>> works if called > > > > >>>>> SIP end point responds with "180 Ringing", but doesn't work if > > > > >>>>> the endpoint responds with early media. > > > > >>>>> > > > > >>>>> _______________________________________________ > > > > >>>>> FreeSWITCH-users mailing list > > > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > > > > > >>>>>us e rs http://www.freeswitch.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-us > > > > > > >>>er s http://www.freeswitch.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-user > > > > > > >s http://www.freeswitch.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 > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.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 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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/20100613/beb28953/attachment-0001.html From codeghar at gmail.com Sun Jun 13 14:36:25 2010 From: codeghar at gmail.com (Code Ghar) Date: Sun, 13 Jun 2010 16:36:25 -0500 Subject: [Freeswitch-users] leg_timeout isn't working. In-Reply-To: References: Message-ID: Hi Nagalenoj Did you try as well as and the latter worked but not the first? I don't have the answer yet but would like to figure out what exactly you did and see if I can replicate it at my end. On Fri, Jun 11, 2010 at 8:09 AM, Nagalenoj H. wrote: > Dear friends, > I've posted an issue in jira(2 days back) but I didn't get any response > there. I just want to confirm whether any one else is facing the same and it > is really an issue. > > Description: > When I tried to execute 'bridge [leg_timeout=10]user/1010', it doesn't quit > ringing if the callee didn't respond in 10 seconds. But when I use it in > {leg_timeout=10}user/1010, it's working. > When I refer the wiki, it is given as it shouldn't be used in curly braces. > > > 'Can be used in per-leg [], but not in global {} for the dialstring.' > -- From wiki ( > http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout) > > Attached the log here, > http://jira.freeswitch.org/browse/MODAPP-433 > > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/5a14e3cb/attachment.html From sos at sokhapkin.dyndns.org Sun Jun 13 14:51:55 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 13 Jun 2010 17:51:55 -0400 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131456.38197.sos@sokhapkin.dyndns.org> Message-ID: <201006131751.55576.sos@sokhapkin.dyndns.org> Thanks! bypass_media=true in combination with ignore_early_media=true did the trick. On Sunday 13 June 2010, Code Ghar wrote: > I tested it the following way: > > > > > > > > > > > > > > > > After 10 seconds, FS sends "cancel" to gateway1 and sends "invite" to > gateway2 after sleeping for a while (optional). Once it sends invite to > gateway2, after 15 seconds FS sends "cancel" to gateway2 and sends "invite" > to gateway3. If after 30 seconds call is not answered it sends "cancel" to > gateway3 and then does not call any other gateway. From all three gateways > I received "183 Session Progress". > > If at any point call is answered, then subsequent gateways are not tried. > For example, if call is answered through gateway1, then after hangup it > doesn't attempt gateway2 and gateway3. > > Now comes your situation, where you need to try one gateway only. In this > situation the following may be of help. > > > > > > > > > > > As you can see, we have eliminated all gateways except gateway1. I have > tested this myself and after 10 seconds FS sends "cancel" to gateway1 and > call is disconnected. > > Hope this helps. I used the examples at following wiki links to create and > test the above configurations. > > Misc. Dialplan Tools bridge > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge > > Dialplan FollowMe > http://wiki.freeswitch.org/wiki/Dialplan_FollowMe > > > > On Sun, Jun 13, 2010 at 1:56 PM, Sergey Okhapkin > > wrote: > > Just tried, leg_timeout doesn't work too. > > > > On Sunday 13 June 2010, Peder wrote: > > > How about leg_timeout on that specific leg of the call. I had an issue > > > where a general call_timeout didn't work if there was no response but > > > leg_timeout did work. Not sure exactly how it works if there is a > > > > response > > > > > though. Worth a shot. > > > > > > -----Original Message----- > > > From: freeswitch-users-bounces at lists.freeswitch.org > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > > > Sergey > > > > > Okhapkin > > > Sent: Sunday, June 13, 2010 1:25 PM > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] How to limit call attemp time? > > > > > > Are you getting SIP 180 or SIP 183 from the B leg? Everything works > > > fine when > > > SIP 180 is received, but call_timeout doesn't work when SIP 183 is > > > received. > > > > > > switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER > > > condition, "media ready" includes early media also :-( > > > > > > On Sunday 13 June 2010, Madovsky wrote: > > > > Sergey, > > > > > > > > I did this > > > > > > > > > > > > repk > > > > > > > > > > > > before bridge and works for me > > > > > > > > Franck > > > > > > > > ----- Original Message ----- > > > > From: "Sergey Okhapkin" > > > > To: > > > > Sent: Sunday, June 13, 2010 2:05 PM > > > > Subject: Re: [Freeswitch-users] How to limit call attemp time? > > > > > > > > > > > > Correct, I need to limit ringing time and continue dialplan execution > > > > if > > > > > no > > > > > > > answer within N seconds. > > > > > > > > It's trivial "find me" service, serial DID forwarding to multiple > > > > PSTN numbers. > > > > > > > > On Sunday 13 June 2010, David Ponzone wrote: > > > > > What you want is a way to limit the ringing time isnt it ? > > > > > I think it's not possible. I needed that some time ago, and I never > > > > > found out the solution. > > > > > > > > > > David Ponzone Direction Technique > > > > > email: david.ponzone at ipeva.fr > > > > > tel: 01 74 03 18 97 > > > > > gsm: 06 66 98 76 34 > > > > > > > > > > Service Client IPeva > > > > > tel: 0811 46 26 26 > > > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > > > ?tablis ? l'intention exclusive de ses destinataires. Toute > > > > > utilisation ou diffusion non autoris?e est interdite. Tout message > > > > > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > > > > > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > > > > > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de > > > > > le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > > > > > > Le 13/06/2010 ? 18:25, Sergey Okhapkin a ?crit : > > > > > > progress_timeout exits the bridge if NO 180 or 183 received, I'm > > > > > > receiving > > > > > > 183. > > > > > > > > > > > > On Sunday 13 June 2010, Jo?o Mesquita wrote: > > > > > >> Look at leg_progress_timeout or just progress_timeout. > > > > > >> > > > > > >> On Sunday, June 13, 2010, Sergey Okhapkin > > > > > >> > > > > > >> wrote: > > > > > >>> Yes, I tried ignore_early_media. call_timeout doesn't work if > > > > > >>> early media > > > > > >>> received. > > > > > >>> > > > > > >>> On Sunday 13 June 2010, Code Ghar wrote: > > > > > >>>> Hi Sergey > > > > > >>>> > > > > > >>>> Have you tried ignore_early_media? It may help. You can find > > > > more > > > > > > > >>>> information from > > > > http://wiki.freeswitch.org/wiki/Channel_Variables > > > > > > > >>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > > > > > >>>> > > > > > >>>> wrote: > > > > > >>>>> Which variable should be set to exit bridge application if > > > > > >>>>> the call > > > > > >>>>> has not been answered within the specified time? call_timeout > > > > > >>>>> variable > > > > > >>>>> works if called > > > > > >>>>> SIP end point responds with "180 Ringing", but doesn't work > > > > > >>>>> if the endpoint responds with early media. > > > > > >>>>> > > > > > >>>>> _______________________________________________ > > > > > >>>>> FreeSWITCH-users mailing list > > > > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > > > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > > > > > > > >>>>>us e rs http://www.freeswitch.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-us > > > > > > > > >>>er s http://www.freeswitch.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-user > > > > > > > > >s http://www.freeswitch.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 > > > > > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.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-user > > >s http://www.freeswitch.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-user > > >s http://www.freeswitch.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 > From sos at sokhapkin.dyndns.org Sun Jun 13 15:02:03 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 13 Jun 2010 18:02:03 -0400 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: <201006131751.55576.sos@sokhapkin.dyndns.org> References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131751.55576.sos@sokhapkin.dyndns.org> Message-ID: <201006131802.04030.sos@sokhapkin.dyndns.org> However I think it's FS bug. call_timeout should wait for answer, regardless of early media. On Sunday 13 June 2010, Sergey Okhapkin wrote: > Thanks! bypass_media=true in combination with ignore_early_media=true did > the trick. > > On Sunday 13 June 2010, Code Ghar wrote: > > I tested it the following way: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > After 10 seconds, FS sends "cancel" to gateway1 and sends "invite" to > > gateway2 after sleeping for a while (optional). Once it sends invite to > > gateway2, after 15 seconds FS sends "cancel" to gateway2 and sends > > "invite" to gateway3. If after 30 seconds call is not answered it sends > > "cancel" to gateway3 and then does not call any other gateway. From all > > three gateways I received "183 Session Progress". > > > > If at any point call is answered, then subsequent gateways are not tried. > > For example, if call is answered through gateway1, then after hangup it > > doesn't attempt gateway2 and gateway3. > > > > Now comes your situation, where you need to try one gateway only. In this > > situation the following may be of help. > > > > > > > > > > > > > > > > > > > > > > As you can see, we have eliminated all gateways except gateway1. I have > > tested this myself and after 10 seconds FS sends "cancel" to gateway1 and > > call is disconnected. > > > > Hope this helps. I used the examples at following wiki links to create > > and test the above configurations. > > > > Misc. Dialplan Tools bridge > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge > > > > Dialplan FollowMe > > http://wiki.freeswitch.org/wiki/Dialplan_FollowMe > > > > > > > > On Sun, Jun 13, 2010 at 1:56 PM, Sergey Okhapkin > > > > wrote: > > > Just tried, leg_timeout doesn't work too. > > > > > > On Sunday 13 June 2010, Peder wrote: > > > > How about leg_timeout on that specific leg of the call. I had an > > > > issue where a general call_timeout didn't work if there was no > > > > response but leg_timeout did work. Not sure exactly how it works if > > > > there is a > > > > > > response > > > > > > > though. Worth a shot. > > > > > > > > -----Original Message----- > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > > > > > Sergey > > > > > > > Okhapkin > > > > Sent: Sunday, June 13, 2010 1:25 PM > > > > To: freeswitch-users at lists.freeswitch.org > > > > Subject: Re: [Freeswitch-users] How to limit call attemp time? > > > > > > > > Are you getting SIP 180 or SIP 183 from the B leg? Everything works > > > > fine when > > > > SIP 180 is received, but call_timeout doesn't work when SIP 183 is > > > > received. > > > > > > > > switch_ivr_originate.c checks MEDIA READY condition, but not for > > > > ANSWER condition, "media ready" includes early media also :-( > > > > > > > > On Sunday 13 June 2010, Madovsky wrote: > > > > > Sergey, > > > > > > > > > > I did this > > > > > > > > > > > > > > > repk > > > > > > > > > > > > > > > before bridge and works for me > > > > > > > > > > Franck > > > > > > > > > > ----- Original Message ----- > > > > > From: "Sergey Okhapkin" > > > > > To: > > > > > Sent: Sunday, June 13, 2010 2:05 PM > > > > > Subject: Re: [Freeswitch-users] How to limit call attemp time? > > > > > > > > > > > > > > > Correct, I need to limit ringing time and continue dialplan > > > > > execution > > > > > > if > > > > > > > no > > > > > > > > > answer within N seconds. > > > > > > > > > > It's trivial "find me" service, serial DID forwarding to multiple > > > > > PSTN numbers. > > > > > > > > > > On Sunday 13 June 2010, David Ponzone wrote: > > > > > > What you want is a way to limit the ringing time isnt it ? > > > > > > I think it's not possible. I needed that some time ago, and I > > > > > > never found out the solution. > > > > > > > > > > > > David Ponzone Direction Technique > > > > > > email: david.ponzone at ipeva.fr > > > > > > tel: 01 74 03 18 97 > > > > > > gsm: 06 66 98 76 34 > > > > > > > > > > > > Service Client IPeva > > > > > > tel: 0811 46 26 26 > > > > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > > > > ?tablis ? l'intention exclusive de ses destinataires. Toute > > > > > > utilisation ou diffusion non autoris?e est interdite. Tout > > > > > > message ?lectronique est susceptible d'alt?ration. IPeva d?cline > > > > > > toute responsabilit? au titre de ce message s'il a ?t? alt?r?, > > > > > > d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > > > > > > message, merci de le d?truire imm?diatement et d'avertir > > > > > > l'exp?diteur. > > > > > > > > > > > > Le 13/06/2010 ? 18:25, Sergey Okhapkin a ?crit : > > > > > > > progress_timeout exits the bridge if NO 180 or 183 received, > > > > > > > I'm receiving > > > > > > > 183. > > > > > > > > > > > > > > On Sunday 13 June 2010, Jo?o Mesquita wrote: > > > > > > >> Look at leg_progress_timeout or just progress_timeout. > > > > > > >> > > > > > > >> On Sunday, June 13, 2010, Sergey Okhapkin > > > > > > >> > > > > > > >> wrote: > > > > > > >>> Yes, I tried ignore_early_media. call_timeout doesn't work if > > > > > > >>> early media > > > > > > >>> received. > > > > > > >>> > > > > > > >>> On Sunday 13 June 2010, Code Ghar wrote: > > > > > > >>>> Hi Sergey > > > > > > >>>> > > > > > > >>>> Have you tried ignore_early_media? It may help. You can find > > > > > > more > > > > > > > > > >>>> information from > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables > > > > > > > > > >>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > > > > > > >>>> > > > > > > >>>> wrote: > > > > > > >>>>> Which variable should be set to exit bridge application if > > > > > > >>>>> the call > > > > > > >>>>> has not been answered within the specified time? > > > > > > >>>>> call_timeout variable > > > > > > >>>>> works if called > > > > > > >>>>> SIP end point responds with "180 Ringing", but doesn't work > > > > > > >>>>> if the endpoint responds with early media. > > > > > > >>>>> > > > > > > >>>>> _______________________________________________ > > > > > > >>>>> FreeSWITCH-users mailing list > > > > > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > > > > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-use > > > > > > >>>>>rs > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > > > > > > > > > >>>>>us e rs http://www.freeswitch.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-us > > > > > > > > > > >>>er s http://www.freeswitch.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-us > > > >er > > > > > > > > > > >s http://www.freeswitch.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 > > > > > > > > > > > > > > > _______________________________________________ > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > > > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > http://www.freeswitch.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-us > > > >er s http://www.freeswitch.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-us > > > >er s http://www.freeswitch.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-user > > >s http://www.freeswitch.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 > From djbinter at gmail.com Sun Jun 13 16:08:49 2010 From: djbinter at gmail.com (DJB International) Date: Sun, 13 Jun 2010 16:08:49 -0700 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: <4C148B7D.6020208@xpirio.com> References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> Message-ID: I experienced the same problem. It also happened to Remote-Party-ID. I have opened Jira for this issue and 2 more related issues as well: http://jira.freeswitch.org/browse/SFSIP-214. You can post your log to the same Jira if you like since I believe it's the similar issue. -djbinter 2010/6/13 Christian L?schenkohl > hello > > the problem is the following > > fs sends in actual git head > P-Asserted-Identity: "43720570500" . > > the git version before (around 7 june) did send (with the same > config/server) > P-Asserted-Identity: "43720570500" > >. > > so the call does not get answered > > pastebin of the two 200 ok packets - http://pastebin.freeswitch.org/13175 > fyi: the system at 93.185.139.77 is a sonus switch > > br > > On 2010-06-12 10:34, Christian L?schenkohl wrote: > > > hello > > > > i expirience a big problem with the latest version (pulled from git). > > inbound sip calls don't get answered (i do answer them with the answer > app). > > does anybody expirience the same issue? > > > > br > > > > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/aab05515/attachment.html From brian at freeswitch.org Sun Jun 13 16:38:38 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 13 Jun 2010 18:38:38 -0500 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> Message-ID: <734A7372-F0BF-4B74-9087-168784BA9008@freeswitch.org> it would work fine if you apply it to the profile and NOT the gateway. /b Sent from my iPad On Jun 13, 2010, at 6:08 PM, DJB International wrote: > I experienced the same problem. It also happened to Remote-Party-ID. > > I have opened Jira for this issue and 2 more related issues as well: http://jira.freeswitch.org/browse/SFSIP-214. > > You can post your log to the same Jira if you like since I believe it's the similar issue. > > > -djbinter > > > > 2010/6/13 Christian L?schenkohl > hello > > the problem is the following > > fs sends in actual git head > P-Asserted-Identity: "43720570500" . > > the git version before (around 7 june) did send (with the same config/server) > P-Asserted-Identity: "43720570500" . > > so the call does not get answered > > pastebin of the two 200 ok packets - http://pastebin.freeswitch.org/13175 > fyi: the system at 93.185.139.77 is a sonus switch > > br > > On 2010-06-12 10:34, Christian L?schenkohl wrote: > > > hello > > > > i expirience a big problem with the latest version (pulled from git). > > inbound sip calls don't get answered (i do answer them with the answer app). > > does anybody expirience the same issue? > > > > br > > > > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100613/7b3106f7/attachment.html From brian at freeswitch.org Sun Jun 13 17:08:58 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 13 Jun 2010 19:08:58 -0500 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> Message-ID: <952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> The issue was we override the cid_type with tech_pvt->cid_type it seems but we already default it to the profile setting which should win if you set it. Patch pushed but it would have worked already if you export sip_cid_type variable before a bridge, Also your use of a gateway is not needed it seems but to each his own. /b On Jun 13, 2010, at 6:08 PM, DJB International wrote: > I experienced the same problem. It also happened to Remote-Party-ID. > > I have opened Jira for this issue and 2 more related issues as well: http://jira.freeswitch.org/browse/SFSIP-214. > > You can post your log to the same Jira if you like since I believe it's the similar issue. > > > -djbinter > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/177cfc03/attachment.html From sean at obscuradigital.com Sun Jun 13 18:14:11 2010 From: sean at obscuradigital.com (Sean Holt) Date: Sun, 13 Jun 2010 18:14:11 -0700 Subject: [Freeswitch-users] 4 second delay Message-ID: Hello list, I?ve been dealing with a particular issue with in-coming calls. Leg A calls into the office, then Leg B (endpoint) picks up call but hears nothing on other side. Wait 4 sec call completes and Leg B can hear the other person. I have Centos 5.4 Latest git build Polycom phones Calling out is not a problem. Not sure how to troubleshoot this issue or maybe there?s a delay setting in the sip profile that waits for the channel to complete. Thanks for the help Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/a15f5adf/attachment.html From djbinter at gmail.com Sun Jun 13 18:48:32 2010 From: djbinter at gmail.com (DJB International) Date: Sun, 13 Jun 2010 18:48:32 -0700 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: <952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> <952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> Message-ID: Brian, Pastebin: http://pastebin.freeswitch.org/13178 git-a95fa59 2010-06-13 19-17-52 -0500 The is now working on sip profile (line 144 - P-Asserted-Identity: "6265101611" .) ; however, the null value is still showing (line 251, line 305, line 365 - Remote-Party-ID: "6628888888" ;party=calling;privacy=off;screen=no. ) Thank you. -djbinter On Sun, Jun 13, 2010 at 5:08 PM, Brian West wrote: > The issue was we override the cid_type with tech_pvt->cid_type it seems but > we already default it to the profile setting which should win if you set it. > > > Patch pushed but it would have worked already if you export sip_cid_type > variable before a bridge, Also your use of a gateway is not needed it seems > but to each his own. > > /b > > > > On Jun 13, 2010, at 6:08 PM, DJB International wrote: > > I experienced the same problem. It also happened to Remote-Party-ID. > > I have opened Jira for this issue and 2 more related issues as well: > http://jira.freeswitch.org/browse/SFSIP-214. > > You can post your log to the same Jira if you like since I believe it's the > similar issue. > > > -djbinter > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/95c650b1/attachment-0001.html From brian at freeswitch.org Sun Jun 13 18:58:37 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 13 Jun 2010 20:58:37 -0500 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> <952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> Message-ID: Yes a jira already exists for that but I don't see how that is even possible to have the domain null in the rpid because the code tries very hard to not have a null rpid_domain in do_invite in sofia_glue. /b On Jun 13, 2010, at 8:48 PM, DJB International wrote: > Brian, > > Pastebin: http://pastebin.freeswitch.org/13178 > > git-a95fa59 2010-06-13 19-17-52 -0500 > > The is now working on sip profile (line 144 - P-Asserted-Identity: "6265101611" .) ; > > however, the null value is still showing (line 251, line 305, line 365 - Remote-Party-ID: "6628888888" ;party=calling;privacy=off;screen=no. ) > > > Thank you. > > -djbinter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/6381a512/attachment.html From djbinter at gmail.com Sun Jun 13 19:12:32 2010 From: djbinter at gmail.com (DJB International) Date: Sun, 13 Jun 2010 19:12:32 -0700 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> <952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> Message-ID: Brian, Do you mean the jira that I opened: SFSIP-214? If so, I think it was closed. Should I re-open it? Thank you, -djbinter On Sun, Jun 13, 2010 at 6:58 PM, Brian West wrote: > Yes a jira already exists for that but I don't see how that is even > possible to have the domain null in the rpid because the code tries very > hard to not have a null rpid_domain in do_invite in sofia_glue. > > /b > > On Jun 13, 2010, at 8:48 PM, DJB International wrote: > > Brian, > > Pastebin: http://pastebin.freeswitch.org/13178 > > git-a95fa59 2010-06-13 19-17-52 -0500 > > The is now working on sip > profile (line 144 - P-Asserted-Identity: "6265101611" 204.110.12.103>.) ; > > however, the null value is still showing (line 251, line 305, line 365 - Remote-Party-ID: > "6628888888" ;party=calling;privacy=off;screen=no. ) > > > > Thank you. > > -djbinter > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/6af2f94f/attachment.html From mike at jerris.com Sun Jun 13 19:19:35 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 13 Jun 2010 22:19:35 -0400 Subject: [Freeswitch-users] odbc on windows In-Reply-To: References: Message-ID: <7FF1B94E-205A-4931-B09B-06259AD7B262@jerris.com> This appears to be not an issue with odbc on windows, but the sql statements we create from the core db not being compatible somehow with ms sql server. Shouldn't BEGIN syntax for transactions be fine for mssql ? Mike On Jun 6, 2010, at 5:41 AM, babak yakhchali wrote: > Hi > I configured my sofia profile internal to use odbc. it connects and creates tables but after that I get lots of errors like bellow > these are just a few of them: > > 2010-06-06 14:09:04.000000 [ERR] switch_odbc.c:427 ERR: [BEGIN] > [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Inc > orrect syntax near 'BEGIN'. > ] > 2010-06-06 14:09:04.000000 [ERR] switch_core_sqldb.c:404 SQL ERR [STATE: 42000 C > ODE 8180 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Statement(s) coul > d not be prepared. > ] = From brian at freeswitch.org Sun Jun 13 19:26:03 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 13 Jun 2010 21:26:03 -0500 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> <952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> Message-ID: <5197122D-905A-4539-9136-F730DC6B58C2@freeswitch.org> Nope... /b On Jun 13, 2010, at 9:12 PM, DJB International wrote: > Brian, > > Do you mean the jira that I opened: SFSIP-214? If so, I think it was closed. Should I re-open it? > > Thank you, > -djbinter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/e13482d6/attachment.html From brian at freeswitch.org Sun Jun 13 19:26:57 2010 From: brian at freeswitch.org (Brian West) Date: Sun, 13 Jun 2010 21:26:57 -0500 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> <952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> Message-ID: The issue is outlined in http://jira.freeswitch.org/browse/FSMOD-61 /b On Jun 13, 2010, at 9:12 PM, DJB International wrote: > Brian, > > Do you mean the jira that I opened: SFSIP-214? If so, I think it was closed. Should I re-open it? > > Thank you, > -djbinter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/5857e093/attachment.html From anthony.minessale at gmail.com Sun Jun 13 19:27:32 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 13 Jun 2010 21:27:32 -0500 Subject: [Freeswitch-users] leg_timeout isn't working. In-Reply-To: References: Message-ID: as already stated, the problem was with the user/ channel and a patch was added to git to address it on the aforementioned jira issue which is closed. On Sun, Jun 13, 2010 at 4:36 PM, Code Ghar wrote: > Hi Nagalenoj > > Did you try > > > > as well as > > > > and the latter worked but not the first? I don't have the answer yet but > would like to figure out what exactly you did and see if I can replicate it > at my end. > > > > On Fri, Jun 11, 2010 at 8:09 AM, Nagalenoj H. wrote: > >> Dear friends, >> I've posted an issue in jira(2 days back) but I didn't get any response >> there. I just want to confirm whether any one else is facing the same and it >> is really an issue. >> >> Description: >> When I tried to execute 'bridge [leg_timeout=10]user/1010', it doesn't >> quit ringing if the callee didn't respond in 10 seconds. But when I use it >> in {leg_timeout=10}user/1010, it's working. >> When I refer the wiki, it is given as it shouldn't be used in curly >> braces. >> >> 'Can be used in per-leg [], but not in global {} for the dialstring.' >> -- From wiki ( >> http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout) >> >> Attached the log here, >> http://jira.freeswitch.org/browse/MODAPP-433 >> >> -- >> Regards, >> Nagalenoj H. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100613/58537c1e/attachment-0001.html From djbinter at gmail.com Sun Jun 13 20:04:32 2010 From: djbinter at gmail.com (DJB International) Date: Sun, 13 Jun 2010 20:04:32 -0700 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> <952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> Message-ID: Brian, One thing I make me confused is that if I have in sofia profile, should the FS change both sip message to terminating and originating gateway to use P-Asserted-Identity. However, I only saw the change to terminating gateway to be P-Asserted-Identity, but the message (sip: 180, 200) that FS sends to originating gateway is still using Remote-Party-ID. Thank you, Brian. -djbinter On Sun, Jun 13, 2010 at 7:26 PM, Brian West wrote: > The issue is outlined in http://jira.freeswitch.org/browse/FSMOD-61 > > /b > > On Jun 13, 2010, at 9:12 PM, DJB International wrote: > > Brian, > > Do you mean the jira that I opened: SFSIP-214? If so, I think it was > closed. Should I re-open it? > > Thank you, > -djbinter > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100613/4a56afe5/attachment.html From jan.berger at video24.no Sun Jun 13 23:15:26 2010 From: jan.berger at video24.no (Jan Berger) Date: Mon, 14 Jun 2010 08:15:26 +0200 Subject: [Freeswitch-users] odbc on windows In-Reply-To: <7FF1B94E-205A-4931-B09B-06259AD7B262@jerris.com> References: <7FF1B94E-205A-4931-B09B-06259AD7B262@jerris.com> Message-ID: <13D5C99F831A4DEB9670A55136F108B3@dell9400> Send me the SQL statement. SQL is SQL, but well - Microsoft have never given a crap about standards. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 14. juni 2010 04:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] odbc on windows This appears to be not an issue with odbc on windows, but the sql statements we create from the core db not being compatible somehow with ms sql server. Shouldn't BEGIN syntax for transactions be fine for mssql ? Mike On Jun 6, 2010, at 5:41 AM, babak yakhchali wrote: > Hi > I configured my sofia profile internal to use odbc. it connects and creates tables but after that I get lots of errors like bellow > these are just a few of them: > > 2010-06-06 14:09:04.000000 [ERR] switch_odbc.c:427 ERR: [BEGIN] > [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Inc > orrect syntax near 'BEGIN'. > ] > 2010-06-06 14:09:04.000000 [ERR] switch_core_sqldb.c:404 SQL ERR [STATE: 42000 C > ODE 8180 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Statement(s) coul > d not be prepared. > ] = _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 Sun Jun 13 23:34:26 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Jun 2010 02:34:26 -0400 Subject: [Freeswitch-users] Full NAT bypass solution with STUN/ICE/FS but without TURN: is possible? In-Reply-To: <4C0E0695.1010308@infosecurity.ch> References: <4C0E014A.1000507@infosecurity.ch> <31EED764-8AAD-4CE8-98BD-E6CE44F80162@gmail.com> <4C0E0695.1010308@infosecurity.ch> Message-ID: <3ABE7973-7B58-48A9-94DB-DAA90F66BC35@jerris.com> with freeswitch, in 99% of the situations, just using rport will pretty much work, stun is a bonus in a couple cases. This will get you through most nat. There are a few stuborn fools who it won't work with. But if they can not provide a working sip alg, then I think those devices are pretty much broken at that point, while not technically, in any practical sense. Mike On Jun 8, 2010, at 5:00 AM, Fabio Pietrosanti (naif) wrote: > I am the SIP phone vendor, but still did not have experience with NAT > bypass technologies and so i am asking whether this is a setup that may > be feasible or not from the "protocols" point of view and from FS point > of view. > > I can install a TURN server but if possible i would like to avoid > setting up server-side systems other than FS. > > Fabio > > On 08/06/10 10.53, David Ponzone wrote: >> Fabio, >> >> this sounds like a question to ask to your SIP phones vendor. >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Sun Jun 13 23:36:40 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Jun 2010 02:36:40 -0400 Subject: [Freeswitch-users] Non-Blocking Music on Hold In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast On Jun 9, 2010, at 8:46 AM, Jon Davies wrote: > Hi There > > I'm attempting to port an application I've written in CCXML/VXML to > FreeSwitch/javascript and finding it a very enjoyable experience. I've > hit the limits of the documentation and my knowledge and was wondering > if there were any clever people on the list who can assist me with one > little part. > > My scenario is: > > Make an outbound call and play it MOH > Make a 2nd outbound call > If the 2nd call connects, take the 1st call off MOH and bridge them together > If the 2nd call fails for whatever reason, take the 1st call off MOH > and play it a 'sorry' message. > > My problem is, all my attempts to play MOH to the first call results > in blocking, meaning I dont ever get to originate my 2nd call. I've > tried > > firstcall.streamFile("music.wav"); > > and > > firstcall.execute( "fifo", "myqueue in undef 'music.wav'" ); > > but both block until I hang up. Anyone got any insight as to how I can > achieve this scenario in a non-blocking manner? I'm using javascript. > From mike at jerris.com Sun Jun 13 23:40:10 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Jun 2010 02:40:10 -0400 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: References: Message-ID: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> On Jun 9, 2010, at 9:36 AM, Vitalii Colosov wrote: > Hi, > > I am trying to make 2 freeswitches (bridged) to work both in bypass media mode - currently with no luck. > Would appreciate your advice. > > Let say I have 2 FreeSWITCH servers - one acts as PROXY (in bypass_media_after_bridge mode) and second processes media. In bypass mode or not, FreeSWITCH is never a proxy, it is always a B2BUA. Processing media or not is your only choice in regards to this. > > I make a call from one sip client to another via these 2 servers (let's say G711 is used everywhere). > SIP CLIENT1 -> FS1 (proxy) -> FS2 (media) -> SIP CLIENT2 > > Proxy server (FS1) sends a re-INVITE and excludes itself from the media path after the call was answered (I am using bypass_media_after_bridge). > > So, the media goes like this: SIP CLIENT1 <-> FS2 <-> SIP CLIENT2 > > Now, I want to try to exclude the media server (FS2) as well (just experimenting, but if both clients use same codec, I suppose this can be legitimately done?). > I put bypass_media_after_bridge=true on the media server (I did it in few places in Lua script). > > But as per sip trace, it does not sends any reinvite after the call was answered. > Only FS1 sends reINVITE. There is something wrong in your script? > So, looks like I cannot remove FS2 from media path like this. > > Do you think it is not possible to acheive this kind of configuration so both servers will work in bypass media? There is nothing different here than the first box as far as your explanation goes. This should work fine, and as you showed with FS1, it does. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/1dbebef9/attachment.html From babak.freeswitch at gmail.com Sun Jun 13 23:43:26 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Mon, 14 Jun 2010 11:13:26 +0430 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: <201006131318.03752.errotan@elder.hu> References: <34D6A82A-5F15-48AF-BCA5-7201AD9CDFCE@gmail.com> <201006131318.03752.errotan@elder.hu> Message-ID: Isn't it possible to chage userstate even if I use odbc and I have direct access to db? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/4b1acaef/attachment.html From mike at jerris.com Sun Jun 13 23:47:08 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Jun 2010 02:47:08 -0400 Subject: [Freeswitch-users] updated sqlite3? In-Reply-To: <4C1125D5.1070004@ewetel.de> References: <4C1125D5.1070004@ewetel.de> Message-ID: If your looking to do anything beyond a very small box, I strongly suggest using some other database and odbc instead of sqlite. Mike On Jun 10, 2010, at 1:50 PM, Helmut Kuper wrote: > Hello, > > today I played around with lua scripts using lua-sqlite3 module. I found > that the delete statement of the sqlite3 version (3.1.3?) FS is using is > not able to handle LIMIT in DELETE statements. Also the centos sqlite3 > cli tool (v3.3.6) doesn't do. > > I found that sqlite3 version 3.6.23 does the trick as long as you use > SQLITE_ENABLE_UPDATE_DELETE_LIMIT during compile time. > > So any chance to get FS sqlite sources updated? > > As long as not, can I riskless update FS's sqlite source to 3.6.23 and > recompile FS? From mike at jerris.com Sun Jun 13 23:54:07 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Jun 2010 02:54:07 -0400 Subject: [Freeswitch-users] uuid_media hangs In-Reply-To: <1276001266045-5153498.post@n2.nabble.com> References: <1275837444425-5145657.post@n2.nabble.com> <1275857911630-5146647.post@n2.nabble.com> <1275859376110-5146705.post@n2.nabble.com> <9D00FC46-26D4-4B54-A61F-0BC81191383E@freeswitch.org> <1275862175540-5146803.post@n2.nabble.com> <1276001266045-5153498.post@n2.nabble.com> Message-ID: I know multiple people who have done interop with BT sip services and freeswitch. Not sure if they just disabled timer support all together. If you can confirm for sure this is an rfc violation, please open a jira on this with the rfc section and a text sip trace of what we are doing, at least so we can track the issue. If you later find a fix and can provide a patch, thats even better. Mike On Jun 8, 2010, at 8:47 AM, peely wrote: > > Hi again, > > I've checked with BT, and they are unable to support reinvites with a > Require: timer. > > Reading the RFC, it seems that new transactions hsould not have the Require: > timer but should revert to Supported: timer. > > Could a future build please support reinvites without the Require: timer > header when sesison timers are enabled? > > > > Thanks, > > > > > Neil. > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p5153498.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 infos at madovsky.org Mon Jun 14 00:09:25 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 14 Jun 2010 03:09:25 -0400 Subject: [Freeswitch-users] default SIP registration timeout Message-ID: Is there any xml param for SIP registration timeout ? Thanks F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/b8b42669/attachment.html From helmut.kuper at ewetel.de Mon Jun 14 01:32:05 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 14 Jun 2010 10:32:05 +0200 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: References: <4C114D36.9000909@ewetel.de> <4C1219ED.2040901@ewetel.de> Message-ID: <4C15E905.70807@ewetel.de> Hi Anthony, I did, and I tested it now. Same behavior as my solution: On FS console i did: snom_bind_key 1 on test 2850 85.16.246.6 internal In Snom SIP-log I found: Received from udp:85.16.246.6:5060 at 14/6/2010 10:19:24:249 (608 bytes): MESSAGE sip:2850 at 85.16.245.213:1073 SIP/2.0 Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKjtNBF9mSHHctK Max-Forwards: 70 From: ;tag=ej91NX74DXHrg To: Call-ID: 4f1db2f1-f230-122d-d0aa-00144fe6e332 CSeq: 132133430 MESSAGE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-17097:17188M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/x-buttons Content-Length: 18 k=11 c=on l=test Sent to udp:85.16.246.6:5060 at 14/6/2010 10:19:24:351 (254 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 85.16.246.6;rport=5060;branch=z9hG4bKjtNBF9mSHHctK From: ;tag=ej91NX74DXHrg To: Call-ID: 4f1db2f1-f230-122d-d0aa-00144fe6e332 CSeq: 132133430 MESSAGE Content-Length: 0 But no lights on on the phones fkeys I use currently FW 8.4.9 and 8.2.25 on snom 370. On 11.06.2010 20:25, Anthony Minessale wrote: > read my last post > > On Fri, Jun 11, 2010 at 6:11 AM, Helmut Kuper > wrote: > regards Helmut From brian at freeswitch.org Mon Jun 14 06:47:49 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Jun 2010 08:47:49 -0500 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> <952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> Message-ID: You can't control what is sent to you with a setting on the sofia profile. /b On Jun 13, 2010, at 10:04 PM, DJB International wrote: > Brian, > > One thing I make me confused is that if I have in sofia profile, should the FS change both sip message to terminating and originating gateway to use P-Asserted-Identity. However, I only saw the change to terminating gateway to be P-Asserted-Identity, but the message (sip: 180, 200) that FS sends to originating gateway is still using Remote-Party-ID. > > Thank you, Brian. > > -djbinter > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/e7e2ea05/attachment.html From jeff at jefflenk.com Mon Jun 14 06:48:12 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 14 Jun 2010 06:48:12 -0700 (PDT) Subject: [Freeswitch-users] odbc on windows In-Reply-To: <13D5C99F831A4DEB9670A55136F108B3@dell9400> References: <7FF1B94E-205A-4931-B09B-06259AD7B262@jerris.com> <13D5C99F831A4DEB9670A55136F108B3@dell9400> Message-ID: <1276523292107-5177504.post@n2.nabble.com> I believe that Transact-SQL requires BEGIN TRANSACTION and COMMIT TRANSACTION http://msdn.microsoft.com/en-us/library/ms188929(v=SQL.105).aspx -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/odbc-on-windows-tp5145042p5177504.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Jun 14 06:50:36 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Jun 2010 08:50:36 -0500 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: <4C15E905.70807@ewetel.de> References: <4C114D36.9000909@ewetel.de> <4C1219ED.2040901@ewetel.de> <4C15E905.70807@ewetel.de> Message-ID: <68625177-D458-4C2E-A0E9-3B01D2D51610@freeswitch.org> Subject just has to be added to the event that is sent. /b On Jun 14, 2010, at 3:32 AM, Helmut Kuper wrote: > Hi Anthony, > > I did, and I tested it now. Same behavior as my solution: > > On FS console i did: From abu.4000 at gmail.com Mon Jun 14 04:27:09 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Mon, 14 Jun 2010 16:57:09 +0530 Subject: [Freeswitch-users] position of members in the fifo Message-ID: Dear all, I am working in fifo using outbound sockets in that my requirement is to say the queue number for the waiting customer for every '5' seconds, I tried this with fifo_position variable but the issue is when ever the customer calls has been answered and hangup then also the queue number is not decremented , is it a correct behaviour ? I can do this using fifo list and fifo list_verbose by parsing its xml structure , but I want to know whether we have a easy way to achieve this. Thanks in advance ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/cd017eea/attachment-0001.html From ravriel_1 at yahoo.com Mon Jun 14 05:45:04 2010 From: ravriel_1 at yahoo.com (Ron Avriel) Date: Mon, 14 Jun 2010 05:45:04 -0700 (PDT) Subject: [Freeswitch-users] Anti-tromboning in FreeSWITCH? In-Reply-To: References: <313308.97850.qm@web45211.mail.sp1.yahoo.com> <101954.30288.qm@web45203.mail.sp1.yahoo.com> <201006091457.26143.sos@sokhapkin.dyndns.org> Message-ID: <683449.69359.qm@web45212.mail.sp1.yahoo.com> Hi Anthony, Thanks for answer. It is possible to achieve the opposite? So that phone1 ----> |FS1| ----> |FS2| phone2 <---- |FS1| <----- |FS2| becomes phone1 -----> |FS1| phone2 <----- |FS1| (modified from your previous answer http://www.mail-archive.com/freeswitch-dev at lists.freeswitch.org/msg02559.html) Thanks, Ron ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, June 9, 2010 10:07:26 PM Subject: Re: [Freeswitch-users] Anti-tromboning in FreeSWITCH? if it detects a bridge in this trombone scenario it will use REFER to transfer the 2 legs on the far end box to be bridged like an attended transfer. On Wed, Jun 9, 2010 at 1:57 PM, Sergey Okhapkin wrote: What sip_auto_simplify settings does? > > >>On Wednesday 09 June 2010, Anthony Minessale wrote: >>> try having sip_auto_simplify channel var set to true at the time of the >>> bridge. >>> >>> On Wed, Jun 9, 2010 at 1:06 PM, Ron Avriel wrote: >>> > Bypass media only connects media of a single bridged call. >>> > The problem here is how to connect media between two different calls. >>> > >>> > Ron >>> > >>> > ------------------------------ >>> > *From:* Milena >>> > *To:* freeswitch-users at lists.freeswitch.org >>> > *Sent:* Wed, June 9, 2010 6:17:26 PM >>> > *Subject:* Re: [Freeswitch-users] Anti-tromboning in FreeSWITCH? >>> > >>> > http://wiki.freeswitch.org/wiki/Bypass_Media >>> > >>> > 2010/6/9 Ron Avriel >>> > >>> >> Hi, >>> >> >>> >> Is there any way to implement Anti-tromboning/Anti-Hairpinning/Media >>> >> Release (http://en.wikipedia.org/wiki/Anti-tromboning) in FreeSWITCH? >>> >> >>> >> My scenario is similar to image in link above. Currently, when user A >>> >> calls B I get two calls and media passing twice through FS. >>> >> Is there any way for media to pass directly between A and B? >>> >> >>> >> Thanks, >>> >> Ron >>> >> >>> >> >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.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 >>> > > >>_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at 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/20100614/a2773b42/attachment-0001.html From helmut.kuper at ewetel.de Mon Jun 14 07:04:13 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 14 Jun 2010 16:04:13 +0200 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: <68625177-D458-4C2E-A0E9-3B01D2D51610@freeswitch.org> References: <4C114D36.9000909@ewetel.de> <4C1219ED.2040901@ewetel.de> <4C15E905.70807@ewetel.de> <68625177-D458-4C2E-A0E9-3B01D2D51610@freeswitch.org> Message-ID: <4C1636DD.3030708@ewetel.de> Hello brian, yes indeed. But since my message is identical to your message except that mine has a subject header, I guess that your way does not work as well. I send my problem in parallel to Snom. They said, that the 8 experience FWs ;) doesn't handle this correct, but the 7.3.30 fw does. They will work on a patch. Have to test that here. Down#t want to downgrade, but hey ... regards Helmut On 14.06.2010 15:50, Brian West wrote: > Subject just has to be added to the event that is sent. > > /b > > On Jun 14, 2010, at 3:32 AM, Helmut Kuper wrote: > >> Hi Anthony, >> >> I did, and I tested it now. Same behavior as my solution: >> >> On FS console i did: > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 vetali100 at gmail.com Mon Jun 14 07:05:27 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Mon, 14 Jun 2010 17:05:27 +0300 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> References: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> Message-ID: Hi Michael, Thanks for important note. Why can't we name FreeSWITCH in Bypass Media mode as PROXY? Which thing it is missing? Regarding the script, I don't think it is incorrect. Most probably the FreeSWITCH is not able to handle such crazy scenario. Because the first reinvite occurs right after the call was answered. So when the second reinvite should happen? Right after the first re-invite? Without any delay? Or maybe it tries to execute both reinvites after the call was answered, and selects the first one? Vitalie 2010/6/14 Michael Jerris > > On Jun 9, 2010, at 9:36 AM, Vitalii Colosov wrote: > > Hi, > > I am trying to make 2 freeswitches (bridged) to work both in bypass media > mode - currently with no luck. > Would appreciate your advice. > > Let say I have 2 FreeSWITCH servers - one acts as PROXY (in > bypass_media_after_bridge mode) and second processes media. > > > In bypass mode or not, FreeSWITCH is never a proxy, it is always a B2BUA. > Processing media or not is your only choice in regards to this. > > > I make a call from one sip client to another via these 2 servers (let's say > G711 is used everywhere). > SIP CLIENT1 -> FS1 (proxy) -> FS2 (media) -> SIP CLIENT2 > > Proxy server (FS1) sends a re-INVITE and excludes itself from the media > path after the call was answered (I am using bypass_media_after_bridge). > > So, the media goes like this: SIP CLIENT1 <-> FS2 <-> SIP CLIENT2 > > Now, I want to try to exclude the media server (FS2) as well (just > experimenting, but if both clients use same codec, I suppose this can be > legitimately done?). > I put bypass_media_after_bridge=true on the media server (I did it in few > places in Lua script). > > *But as per sip trace, it does not sends any reinvite after the call was > answered.* > *Only FS1 sends reINVITE.* > > > There is something wrong in your script? > > So, looks like I cannot remove FS2 from media path like this. > > Do you think it is not possible to acheive this kind of configuration so > both servers will work in bypass media? > > > There is nothing different here than the first box as far as your > explanation goes. This should work fine, and as you showed with FS1, it > does. > > Mike > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/8f8866e2/attachment.html From brian at freeswitch.org Mon Jun 14 07:31:40 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Jun 2010 09:31:40 -0500 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: <4C1636DD.3030708@ewetel.de> References: <4C114D36.9000909@ewetel.de> <4C1219ED.2040901@ewetel.de> <4C15E905.70807@ewetel.de> <68625177-D458-4C2E-A0E9-3B01D2D51610@freeswitch.org> <4C1636DD.3030708@ewetel.de> Message-ID: <4C987513-D69F-4557-B1B5-48AC68731AFD@freeswitch.org> Well the subject requirement must be new because it was working fine. /b On Jun 14, 2010, at 9:04 AM, Helmut Kuper wrote: > Hello brian, > > yes indeed. But since my message is identical to your message except > that mine has a subject header, I guess that your way does not work as well. > > I send my problem in parallel to Snom. They said, that the 8 experience > FWs ;) doesn't handle this correct, but the 7.3.30 fw does. They will > work on a patch. > > Have to test that here. Down#t want to downgrade, but hey ... > > regards > Helmut From brian at freeswitch.org Mon Jun 14 07:31:16 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Jun 2010 09:31:16 -0500 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: References: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> Message-ID: On Jun 14, 2010, at 9:05 AM, Vitalii Colosov wrote: > Hi Michael, > > Thanks for important note. > Why can't we name FreeSWITCH in Bypass Media mode as PROXY? Which thing it is missing? Because when you proxy the media it goes thru your FreeSWITCH instance and when you're in bypass you don't have the media it goes direct P2P /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/466d1c1e/attachment.html From brian at freeswitch.org Mon Jun 14 07:33:07 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Jun 2010 09:33:07 -0500 Subject: [Freeswitch-users] Anti-tromboning in FreeSWITCH? In-Reply-To: <683449.69359.qm@web45212.mail.sp1.yahoo.com> References: <313308.97850.qm@web45211.mail.sp1.yahoo.com> <101954.30288.qm@web45203.mail.sp1.yahoo.com> <201006091457.26143.sos@sokhapkin.dyndns.org> <683449.69359.qm@web45212.mail.sp1.yahoo.com> Message-ID: <047C0947-DB4A-43D1-909C-BDB624669245@freeswitch.org> sip_auto_simplify=true as a global variable it should do exactly that. /b On Jun 14, 2010, at 7:45 AM, Ron Avriel wrote: > Hi Anthony, > > Thanks for answer. It is possible to achieve the opposite? So that > phone1 ----> |FS1| ----> |FS2| > phone2 <---- |FS1| <----- |FS2| > > becomes > > phone1 -----> |FS1| > phone2 <----- |FS1| > > > (modified from your previous answer http://www.mail-archive.com/freeswitch-dev at lists.freeswitch.org/msg02559.html) > > Thanks, > Ron > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/0d75e99c/attachment.html From helmut.kuper at ewetel.de Mon Jun 14 07:43:30 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 14 Jun 2010 16:43:30 +0200 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: <4C1636DD.3030708@ewetel.de> References: <4C114D36.9000909@ewetel.de> <4C1219ED.2040901@ewetel.de> <4C15E905.70807@ewetel.de> <68625177-D458-4C2E-A0E9-3B01D2D51610@freeswitch.org> <4C1636DD.3030708@ewetel.de> Message-ID: <4C164012.6030105@ewetel.de> Hi Brian, ok, got it. mod_snom and also my solution works perfectly with Snom 7.3.29 firmware. Thx for your help. regards helmut On 14.06.2010 16:04, Helmut Kuper wrote: > Hello brian, > > yes indeed. But since my message is identical to your message except > that mine has a subject header, I guess that your way does not work as well. > > I send my problem in parallel to Snom. They said, that the 8 experience > FWs ;) doesn't handle this correct, but the 7.3.30 fw does. They will > work on a patch. > > Have to test that here. Down#t want to downgrade, but hey ... From brian at freeswitch.org Mon Jun 14 07:46:43 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Jun 2010 09:46:43 -0500 Subject: [Freeswitch-users] help about a system with login logout In-Reply-To: References: <34D6A82A-5F15-48AF-BCA5-7201AD9CDFCE@gmail.com> <201006131318.03752.errotan@elder.hu> Message-ID: <0FE92087-4DBA-4245-80F2-073BAB1ABA14@freeswitch.org> NO. /b On Jun 14, 2010, at 1:43 AM, babak yakhchali wrote: > Isn't it possible to chage userstate even if I use odbc and I have direct access to db? From vetali100 at gmail.com Mon Jun 14 08:06:47 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Mon, 14 Jun 2010 18:06:47 +0300 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: References: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> Message-ID: Hi Brian, I did not find the mandatory requirement for the proxy server to proxy also media: http://www.ietf.org/rfc/rfc3261.txt Proxy, Proxy Server: An intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is sent to another entity "closer" to the targeted user. Proxies are also useful for enforcing policy (for example, making sure a user is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it. So I am still not convinced that I cannot name FreeSWITCH in Bypass Media mode as a PROXY :-) Because it does all the things described: 1.intermediary entity 2.plays the role of routing 3.its job is to ensure that a request is sent to another entity "closer" to the targeted user Please pay attention that nothing is written about media here. Vitalie 2010/6/14 Brian West > > On Jun 14, 2010, at 9:05 AM, Vitalii Colosov wrote: > > Hi Michael, > > Thanks for important note. > Why can't we name FreeSWITCH in Bypass Media mode as PROXY? Which thing it > is missing? > > > Because when you proxy the media it goes thru your FreeSWITCH instance and > when you're in bypass you don't have the media it goes direct P2P > > /b > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/d526820c/attachment-0001.html From brian at freeswitch.org Mon Jun 14 08:18:05 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Jun 2010 10:18:05 -0500 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: References: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> Message-ID: <41CFD2E5-5228-4C97-BA3E-8D95D7091205@freeswitch.org> You are clearly overlooking that we aren't a proxy server. We are a b2bua. /b On Jun 14, 2010, at 10:06 AM, Vitalii Colosov wrote: > Hi Brian, > > I did not find the mandatory requirement for the proxy server to proxy also media: > > http://www.ietf.org/rfc/rfc3261.txt > > Proxy, Proxy Server: An intermediary entity that acts as both a > server and a client for the purpose of making requests on > behalf of other clients. A proxy server primarily plays the > role of routing, which means its job is to ensure that a > request is sent to another entity "closer" to the targeted > user. Proxies are also useful for enforcing policy (for > example, making sure a user is allowed to make a call). A > proxy interprets, and, if necessary, rewrites specific parts of > a request message before forwarding it. > > So I am still not convinced that I cannot name FreeSWITCH in Bypass Media mode as a PROXY :-) > Because it does all the things described: > > 1.intermediary entity > 2.plays the role of routing > 3.its job is to ensure that a request is sent to another entity "closer" to the targeted user > > Please pay attention that nothing is written about media here. > > Vitalie > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/0303658d/attachment.html From brian at freeswitch.org Mon Jun 14 08:21:09 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Jun 2010 10:21:09 -0500 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: References: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> Message-ID: <63AAF9EB-F28F-4D71-A659-AC44ED626173@freeswitch.org> You still can't call this a proxy because its just passing the SDP's across so the media goes P2P. /b On Jun 14, 2010, at 10:06 AM, Vitalii Colosov wrote: > So I am still not convinced that I cannot name FreeSWITCH in Bypass Media mode as a PROXY :-) > Because it does all the things described: From kris at kriskinc.com Mon Jun 14 08:27:50 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 14 Jun 2010 11:27:50 -0400 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: References: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> Message-ID: Vitali, You can call anything whatever you want. However, the purpose of any language is to effectively communicate meaning to others. Calling FreeSWITCH a "SIP Proxy" in SIP circles (with SIP people) will only confuse them and make you seem ignorant. When discussing SIP signaling a "proxy" looks very distinct. They basically slap on a Record-Route/Route/Via header and forward the message. FreeSWITCH builds a completely new and separate call leg with rewritten headers for each call leaving the system. This makes it a B2BUA. On Mon, Jun 14, 2010 at 11:06 AM, Vitalii Colosov wrote: > Hi Brian, > I did not find the mandatory requirement for the proxy server to proxy also > media: > http://www.ietf.org/rfc/rfc3261.txt > Proxy, Proxy Server: An intermediary entity that acts as both a > ?? ? ? ? server and a client for the purpose of making requests on > ?? ? ? ? behalf of other clients. ?A proxy server primarily plays the > ?? ? ? ? role of routing, which means its job is to ensure that a > ?? ? ? ? request is sent to another entity "closer" to the targeted > ?? ? ? ? user. ?Proxies are also useful for enforcing policy (for > ?? ? ? ? example, making sure a user is allowed to make a call). ?A > ?? ? ? ? proxy interprets, and, if necessary, rewrites specific parts of > ?? ? ? ? a request message before forwarding it. > So I am still not convinced that I cannot name FreeSWITCH in Bypass Media > mode as a PROXY :-) > Because it does all the things described: > 1.intermediary entity > 2.plays the?role of routing > 3.its job is to ensure that a?request is sent to another entity "closer" to > the targeted?user > Please pay attention that nothing is written about media here. > Vitalie > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From vetali100 at gmail.com Mon Jun 14 08:39:51 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Mon, 14 Jun 2010 18:39:51 +0300 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: References: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> Message-ID: Thank you! Can I name the FreeSWITCH as a powerful B2BUA that contains all SIP Proxy functionality? Vitalie 2010/6/14 Kristian Kielhofner > Vitali, > > You can call anything whatever you want. However, the purpose of > any language is to effectively communicate meaning to others. Calling > FreeSWITCH a "SIP Proxy" in SIP circles (with SIP people) will only > confuse them and make you seem ignorant. > > When discussing SIP signaling a "proxy" looks very distinct. They > basically slap on a Record-Route/Route/Via header and forward the > message. FreeSWITCH builds a completely new and separate call leg > with rewritten headers for each call leaving the system. This makes > it a B2BUA. > > On Mon, Jun 14, 2010 at 11:06 AM, Vitalii Colosov > wrote: > > Hi Brian, > > I did not find the mandatory requirement for the proxy server to proxy > also > > media: > > http://www.ietf.org/rfc/rfc3261.txt > > Proxy, Proxy Server: An intermediary entity that acts as both a > > server and a client for the purpose of making requests on > > behalf of other clients. A proxy server primarily plays the > > role of routing, which means its job is to ensure that a > > request is sent to another entity "closer" to the targeted > > user. Proxies are also useful for enforcing policy (for > > example, making sure a user is allowed to make a call). A > > proxy interprets, and, if necessary, rewrites specific parts of > > a request message before forwarding it. > > So I am still not convinced that I cannot name FreeSWITCH in Bypass Media > > mode as a PROXY :-) > > Because it does all the things described: > > 1.intermediary entity > > 2.plays the role of routing > > 3.its job is to ensure that a request is sent to another entity "closer" > to > > the targeted user > > Please pay attention that nothing is written about media here. > > Vitalie > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/cbb9abc1/attachment.html From anthony.minessale at gmail.com Mon Jun 14 09:13:15 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Jun 2010 11:13:15 -0500 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: References: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> Message-ID: Try bypass_media=true instead. This sets the call in bypass right off the bat with no need for reinvites. On Jun 14, 2010 10:45 AM, "Vitalii Colosov" wrote: Thank you! Can I name the FreeSWITCH as a powerful B2BUA that contains all SIP Proxy functionality? Vitalie 2010/6/14 Kristian Kielhofner > > Vitali, > > You can call anything whatever you want. However, the purpose of > any language i... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/09b5e095/attachment.html From djbinter at gmail.com Mon Jun 14 09:14:31 2010 From: djbinter at gmail.com (DJB International) Date: Mon, 14 Jun 2010 09:14:31 -0700 Subject: [Freeswitch-users] major problem with latest version In-Reply-To: References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com> <952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> Message-ID: Brain, I meant was what sofia sent out. For instance, from the previous pastebin: http://pastebin.freeswitch.org/13178, if you look at line 251, 305, and 365. Call Flow is from 75.214.107.143 (originating) -> 204.110.12.103 (FS) -> 209.9.188.91 (terminating) Thank you, -djbinter On Mon, Jun 14, 2010 at 6:47 AM, Brian West wrote: > You can't control what is sent to you with a setting on the sofia profile. > > /b > > On Jun 13, 2010, at 10:04 PM, DJB International wrote: > > Brian, > > One thing I make me confused is that if I have name="caller-id-type" value="pid"/> in sofia profile, should the FS change > both sip message to terminating and originating gateway to use > P-Asserted-Identity. However, I only saw the change to terminating gateway > to be P-Asserted-Identity, but the message (sip: 180, 200) that FS sends to > originating gateway is still using Remote-Party-ID. > > Thank you, Brian. > > -djbinter > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/7e108294/attachment-0001.html From helmut.kuper at ewetel.de Mon Jun 14 09:15:51 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 14 Jun 2010 18:15:51 +0200 Subject: [Freeswitch-users] LED-Controlling on Snom phones. Missing Subject header In-Reply-To: <4C164012.6030105@ewetel.de> References: <4C114D36.9000909@ewetel.de> <4C1219ED.2040901@ewetel.de> <4C15E905.70807@ewetel.de> <68625177-D458-4C2E-A0E9-3B01D2D51610@freeswitch.org> <4C1636DD.3030708@ewetel.de> <4C164012.6030105@ewetel.de> Message-ID: <4C1655B7.5020405@ewetel.de> Hi Brian, just a new info from Snom: Series 8 works as well. You have to configure snoms fkey to "button" and assign a label aka index to it. The index must be the same as in the "k=" in message body option - and a number!. regards helmut On 14.06.2010 16:43, Helmut Kuper wrote: > Hi Brian, > > > ok, got it. mod_snom and also my solution works perfectly with Snom > 7.3.29 firmware. > > Thx for your help. > > regards > helmut > From infos at madovsky.org Mon Jun 14 09:44:26 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 14 Jun 2010 12:44:26 -0400 Subject: [Freeswitch-users] major problem with latest version References: <4C13469C.5040103@xpirio.com> <4C148B7D.6020208@xpirio.com><952E2CFA-643D-44D7-9FE7-46C71627D57C@freeswitch.org> Message-ID: <6A151EBCBEEC4CACB6CA3E99EAC63D1E@MOBILEE1705> hey Brain, give neurons to FS community ! :D ----- Original Message ----- From: DJB International To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 14, 2010 12:14 PM Subject: Re: [Freeswitch-users] major problem with latest version Brain, I meant was what sofia sent out. For instance, from the previous pastebin: http://pastebin.freeswitch.org/13178, if you look at line 251, 305, and 365. Call Flow is from 75.214.107.143 (originating) -> 204.110.12.103 (FS) -> 209.9.188.91 (terminating) Thank you, -djbinter On Mon, Jun 14, 2010 at 6:47 AM, Brian West wrote: You can't control what is sent to you with a setting on the sofia profile. /b On Jun 13, 2010, at 10:04 PM, DJB International wrote: Brian, One thing I make me confused is that if I have in sofia profile, should the FS change both sip message to terminating and originating gateway to use P-Asserted-Identity. However, I only saw the change to terminating gateway to be P-Asserted-Identity, but the message (sip: 180, 200) that FS sends to originating gateway is still using Remote-Party-ID. Thank you, Brian. -djbinter _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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/20100614/2b7e205e/attachment.html From david.ponzone at gmail.com Mon Jun 14 10:24:30 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 14 Jun 2010 19:24:30 +0200 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: References: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> Message-ID: <9F57D9D5-73F7-4D04-B686-DF1171A759A7@gmail.com> No, it's a B2BUA, with an embedded RTP Proxy. FS can't be a SIP Proxy. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 14/06/2010 ? 17:39, Vitalii Colosov a ?crit : > Thank you! > > Can I name the FreeSWITCH as a powerful B2BUA that contains all SIP > Proxy functionality? > > Vitalie > > > 2010/6/14 Kristian Kielhofner > Vitali, > > You can call anything whatever you want. However, the purpose of > any language is to effectively communicate meaning to others. Calling > FreeSWITCH a "SIP Proxy" in SIP circles (with SIP people) will only > confuse them and make you seem ignorant. > > When discussing SIP signaling a "proxy" looks very distinct. They > basically slap on a Record-Route/Route/Via header and forward the > message. FreeSWITCH builds a completely new and separate call leg > with rewritten headers for each call leaving the system. This makes > it a B2BUA. > > On Mon, Jun 14, 2010 at 11:06 AM, Vitalii Colosov > wrote: > > Hi Brian, > > I did not find the mandatory requirement for the proxy server to > proxy also > > media: > > http://www.ietf.org/rfc/rfc3261.txt > > Proxy, Proxy Server: An intermediary entity that acts as both a > > server and a client for the purpose of making requests on > > behalf of other clients. A proxy server primarily plays > the > > role of routing, which means its job is to ensure that a > > request is sent to another entity "closer" to the targeted > > user. Proxies are also useful for enforcing policy (for > > example, making sure a user is allowed to make a call). A > > proxy interprets, and, if necessary, rewrites specific > parts of > > a request message before forwarding it. > > So I am still not convinced that I cannot name FreeSWITCH in > Bypass Media > > mode as a PROXY :-) > > Because it does all the things described: > > 1.intermediary entity > > 2.plays the role of routing > > 3.its job is to ensure that a request is sent to another entity > "closer" to > > the targeted user > > Please pay attention that nothing is written about media here. > > Vitalie > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100614/720fb1fb/attachment.html From jerry.richards at teotech.com Mon Jun 14 10:24:35 2010 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 14 Jun 2010 10:24:35 -0700 Subject: [Freeswitch-users] Dropped Call During VM Email Delivery? Message-ID: We getting dropped calls recently. They seem to appear in the vicinity where mod_voicemail is delivering a voice mail Email (at least that's where we've seen them). Below, I show a snippet from the Freeswitch log that shows two calls (on extensions 1028 and 1063) getting simultaneously disconnected. Do you know what might be the cause? 2010-06-14 08:59:02.495697 [DEBUG] mod_voicemail.c:2394 Deliver VM to 1036 at 192.168.72.141 2010-06-14 08:59:02.500634 [DEBUG] ozmod_sangoma_boost.c:1174 1:23 STATE [DOWN] 2010-06-14 08:59:02.500634 [WARNING] sangoma_boost_client.c:221 TX EVENT (N): CALL_STOPPED_ACK:(86) [w1g23] Rc=0 CSid=0 Seq=766 2010-06-14 08:59:02.500634 [DEBUG] zap_io.c:1388 channel done 1:23 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/1028 at 192.168.72.141:5060 ending bridge by request from read function 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/1063 at 192.168.72.141:5060 ending bridge by request from read function 2010-06-14 08:59:04.500353 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/1028 at 192.168.72.141:5060 [BREAK] 2010-06-14 08:59:04.500353 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/1063 at 192.168.72.141:5060 [BREAK] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD DONE [sofia/internal/1028 at 192.168.72.141:5060] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:585 Send signal OpenZAP/1:1/18007297580 at g1 [BREAK] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/1032 at 192.168.72.141:5060 ending bridge by request from read function 2010-06-14 08:59:04.500353 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/1032 at 192.168.72.141:5060 [BREAK] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD DONE [sofia/internal/1032 at 192.168.72.141:5060] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:585 Send signal OpenZAP/1:3/19403232644 at g1 [BREAK] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD DONE [sofia/internal/1063 at 192.168.72.141:5060] 2010-06-14 08:59:04.500353 [DEBUG] switch_ivr_bridge.c:585 Send signal OpenZAP/1:2/14253491003 at g1 [BREAK] 2010-06-14 08:59:04.501359 [DEBUG] switch_utils.c:631 Emailed file [/tmp/mail.12765311428da7] to [chuck.wichser at teotech.com] 2010-06-14 08:59:04.501359 [DEBUG] mod_voicemail.c:2562 Sending message to chuck.wichser at teotech.com Best Regards, Jerry From dcolombo at voismart.it Mon Jun 14 08:22:00 2010 From: dcolombo at voismart.it (Davide Colombo) Date: Mon, 14 Jun 2010 17:22:00 +0200 (CEST) Subject: [Freeswitch-users] Question about execute_on_answer Message-ID: <260919296.2294.1276528920082.JavaMail.root@mx.voismart.com> Hi all, i'm trying to use "execute_on_answer" channel variable with a transfer before a bridge: When i call 1019 from 1000 and 1019 answers, i have this situation: 1019 is transfered to myivrpresentation but in console i can see this error 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 sofia/internal/1000 at fs2-devel.voismart.net Restore previous codec PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] while 1000 goes to next action after bridge in the same extension. Is it possible to transfer called channel (when called party answers) to a particular extension and leave caller channel in ringing state? Best Regards From mike at jerris.com Mon Jun 14 11:46:37 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Jun 2010 14:46:37 -0400 Subject: [Freeswitch-users] odbc on windows In-Reply-To: <1276523292107-5177504.post@n2.nabble.com> References: <7FF1B94E-205A-4931-B09B-06259AD7B262@jerris.com> <13D5C99F831A4DEB9670A55136F108B3@dell9400> <1276523292107-5177504.post@n2.nabble.com> Message-ID: <038B6AC2-5A55-4251-A11E-25C7CEF04BD0@jerris.com> can someone make sure there is a bug in jira about this issue. Mike On Jun 14, 2010, at 9:48 AM, Jeff Lenk wrote: > > I believe that Transact-SQL requires BEGIN TRANSACTION and COMMIT TRANSACTION > > http://msdn.microsoft.com/en-us/library/ms188929(v=SQL.105).aspx From msc at freeswitch.org Mon Jun 14 12:32:25 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jun 2010 12:32:25 -0700 Subject: [Freeswitch-users] 4 second delay In-Reply-To: References: Message-ID: Hop on the console, turn on siptrace and watch the call flow for clues. You might need to do a tcpdump capturing both signaling and media and then analyze in Wireshark to see what exactly is happening. -MC On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt wrote: > Hello list, > > I?ve been dealing with a particular issue with in-coming calls. > Leg A calls into the office, then Leg B (endpoint) picks up call but hears > nothing on other side. Wait 4 sec call completes and Leg B can hear the > other person. > > I have Centos 5.4 > Latest git build > Polycom phones > > Calling out is not a problem. > > Not sure how to troubleshoot this issue or maybe there?s a delay setting in > the sip profile that waits for the channel to complete. > > Thanks for the help > Sean > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/c6ab0c53/attachment.html From jeff at jefflenk.com Mon Jun 14 12:40:15 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 14 Jun 2010 12:40:15 -0700 (PDT) Subject: [Freeswitch-users] odbc on windows In-Reply-To: <038B6AC2-5A55-4251-A11E-25C7CEF04BD0@jerris.com> References: <7FF1B94E-205A-4931-B09B-06259AD7B262@jerris.com> <13D5C99F831A4DEB9670A55136F108B3@dell9400> <1276523292107-5177504.post@n2.nabble.com> <038B6AC2-5A55-4251-A11E-25C7CEF04BD0@jerris.com> Message-ID: <1276544415391-5179004.post@n2.nabble.com> Done- http://jira.freeswitch.org/browse/FSCORE-623 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/odbc-on-windows-tp5145042p5179004.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Jun 14 12:43:17 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jun 2010 12:43:17 -0700 Subject: [Freeswitch-users] How to limit call attemp time? In-Reply-To: <201006131802.04030.sos@sokhapkin.dyndns.org> References: <201006131142.55400.sos@sokhapkin.dyndns.org> <201006131751.55576.sos@sokhapkin.dyndns.org> <201006131802.04030.sos@sokhapkin.dyndns.org> Message-ID: On Sun, Jun 13, 2010 at 3:02 PM, Sergey Okhapkin wrote: > However I think it's FS bug. call_timeout should wait for answer, > regardless > of early media. > I'm not sure that statement is correct. The purpose of the call_timeout variable is documented here: http://wiki.freeswitch.org/wiki/Channel_Variables#call_timeout Perhaps you needed this var: http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_answer_timeout According to the description, it sounds exactly like what you were asking about. Also, you can attack this problem from a totally different angle but using execute_on_answer: http://wiki.freeswitch.org/wiki/Variable_execute_on_answer And then you can do whatever you want for 20 seconds (or however long) and if the far end never answers then you can move on and handle the no answer condition. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/b555edec/attachment.html From msc at freeswitch.org Mon Jun 14 12:50:52 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jun 2010 12:50:52 -0700 Subject: [Freeswitch-users] Two bridged FreeSWITCH servers - both in bypass_media mode - possible? In-Reply-To: <9F57D9D5-73F7-4D04-B686-DF1171A759A7@gmail.com> References: <10CC0553-8C72-49E5-B4FF-6020361DF87B@jerris.com> <9F57D9D5-73F7-4D04-B686-DF1171A759A7@gmail.com> Message-ID: On Mon, Jun 14, 2010 at 10:24 AM, David Ponzone wrote: > No, it's a B2BUA, with an embedded RTP Proxy. > FS can't be a SIP Proxy. > David, Well said. People hear the word "proxy" and it's easy to make assumption about exactly *what* is being proxied. So, for posterity's sake: FreeSWITCH is not, and CANNOT, be a SIP proxy. Period. No discussion, no debate. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/50e394a6/attachment.html From msc at freeswitch.org Mon Jun 14 14:46:42 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jun 2010 14:46:42 -0700 Subject: [Freeswitch-users] Question about execute_on_answer In-Reply-To: <260919296.2294.1276528920082.JavaMail.root@mx.voismart.com> References: <260919296.2294.1276528920082.JavaMail.root@mx.voismart.com> Message-ID: I'm not sure I follow what you're trying to accomplish... You call from x1000 to x1019... and what exactly do you want to have happen when 1019 answers? -MC On Mon, Jun 14, 2010 at 8:22 AM, Davide Colombo wrote: > Hi all, > > i'm trying to use "execute_on_answer" channel variable with a transfer > before a bridge: > > > > > When i call 1019 from 1000 and 1019 answers, i have this situation: > > 1019 is transfered to myivrpresentation but in console i can see this error > > 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 sofia/internal/ > 1000 at fs2-devel.voismart.net Restore previous codec PCMU:0. > 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot create > outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] > 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > > while 1000 goes to next action after bridge in the same extension. > > Is it possible to transfer called channel (when called party answers) to a > particular extension and leave caller channel in ringing state? > > > Best Regards > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/199252dd/attachment.html From dcolombo at voismart.it Mon Jun 14 15:12:01 2010 From: dcolombo at voismart.it (Davide Colombo) Date: Tue, 15 Jun 2010 00:12:01 +0200 (CEST) Subject: [Freeswitch-users] Question about execute_on_answer In-Reply-To: <1461100240.2405.1276553519171.JavaMail.root@mx.voismart.com> Message-ID: <725276347.2407.1276553521706.JavaMail.root@mx.voismart.com> Hi, when 1019 answers, i transfer its channel (called channel party) to an IVR context using "execute_on_answer". In this context the called party (1019 in this case) can listen to a message where he can choose to connect/hang-up the calling party using dtmf tones. In the meanwhile i need that calling party remains in a ringing state. Best regards ----- Messaggio originale ----- Da: "Michael Collins" A: freeswitch-users at lists.freeswitch.org Inviato: Luned?, 14 giugno 2010 23:46:42 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer I'm not sure I follow what you're trying to accomplish... You call from x1000 to x1019... and what exactly do you want to have happen when 1019 answers? -MC On Mon, Jun 14, 2010 at 8:22 AM, Davide Colombo < dcolombo at voismart.it > wrote: Hi all, i'm trying to use "execute_on_answer" channel variable with a transfer before a bridge: When i call 1019 from 1000 and 1019 answers, i have this situation: 1019 is transfered to myivrpresentation but in console i can see this error 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 sofia/internal/ 1000 at fs2-devel.voismart.net Restore previous codec PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] while 1000 goes to next action after bridge in the same extension. Is it possible to transfer called channel (when called party answers) to a particular extension and leave caller channel in ringing state? Best Regards _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 From msc at freeswitch.org Mon Jun 14 15:37:18 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jun 2010 15:37:18 -0700 Subject: [Freeswitch-users] Question about execute_on_answer In-Reply-To: <725276347.2407.1276553521706.JavaMail.root@mx.voismart.com> References: <1461100240.2405.1276553519171.JavaMail.root@mx.voismart.com> <725276347.2407.1276553521706.JavaMail.root@mx.voismart.com> Message-ID: Aha! You are trying to do some sort of answer verification, correct? You do know that FS already has this feature, correct? Check it out: http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables Make sure that you aren't duplicating functionality before continuing. -MC On Mon, Jun 14, 2010 at 3:12 PM, Davide Colombo wrote: > Hi, > when 1019 answers, i transfer its channel (called channel party) to an IVR > context using "execute_on_answer". In this context the called party (1019 in > this case) can listen to a message where he can choose to connect/hang-up > the calling party using dtmf tones. > In the meanwhile i need that calling party remains in a ringing state. > > Best regards > > ----- Messaggio originale ----- > Da: "Michael Collins" > A: freeswitch-users at lists.freeswitch.org > Inviato: Luned?, 14 giugno 2010 23:46:42 > Oggetto: Re: [Freeswitch-users] Question about execute_on_answer > > I'm not sure I follow what you're trying to accomplish... > > You call from x1000 to x1019... and what exactly do you want to have > happen when 1019 answers? > > -MC > > > On Mon, Jun 14, 2010 at 8:22 AM, Davide Colombo < dcolombo at voismart.it > > wrote: > > > Hi all, > > i'm trying to use "execute_on_answer" channel variable with a transfer > before a bridge: > > > > > When i call 1019 from 1000 and 1019 answers, i have this situation: > > 1019 is transfered to myivrpresentation but in console i can see this > error > > 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 > sofia/internal/ 1000 at fs2-devel.voismart.net Restore previous codec > PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 > Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot > create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] > 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > > while 1000 goes to next action after bridge in the same extension. > > Is it possible to transfer called channel (when called party answers) to > a particular extension and leave caller channel in ringing state? > > > Best Regards > > _______________________________________________ FreeSWITCH-users mailing > list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/3e294171/attachment.html From msc at freeswitch.org Mon Jun 14 16:03:17 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jun 2010 16:03:17 -0700 Subject: [Freeswitch-users] New Content on freeswitch.org Message-ID: Hello there! Just an FYI, please visit http://www.freeswitch.org and check out the many new posts we have. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100614/13f34731/attachment.html From gmaruzz at celliax.org Mon Jun 14 16:24:20 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 15 Jun 2010 01:24:20 +0200 Subject: [Freeswitch-users] New Content on freeswitch.org In-Reply-To: References: Message-ID: On Tue, Jun 15, 2010 at 1:03 AM, Michael Collins wrote: > Hello there! > > Just an FYI, please visit http://www.freeswitch.org and check out the many > new posts we have. Kewl! -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From nagalenoj at gmail.com Mon Jun 14 22:28:29 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Tue, 15 Jun 2010 10:58:29 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout Message-ID: Dear friends, I've tried using the group_confirm_cancel_timeout channel variable. I've written a testing script to get digits before bridging. But, it doesn't seem to be working. My understanding after reading wiki is, * When I dial [leg_timeout=10]user/1005, if he answers before timeout and in the process of giving digits, then the call shouldn't be disconnected after the leg_timeout secs (10 sec in the example). But, When I experiment it, the call is getting disconnected after 10 seconds and it doesn't bother whether the callee has answered the call(Started giving digits) or not answered at all. I've checked it with nc as follows, sendmsg call-command: execute execute-app-name: set execute-app-arg: group_confirm_key=exec sendmsg call-command: execute execute-app-name: set execute-app-arg: group_confirm_file=perl /root/confirm.pl sendmsg call-command: execute execute-app-name: set execute-app-arg: group_confirm_cancel_timeout=1 sendmsg call-command: execute execute-app-name: bridge execute-app-arg: [leg_timeout=10]user/1005 And here is the script, use freeswitch; our $session; my $digit; while(1) { # Wait till response timeout for the first digit. $digit = $session->getDigits(1, "", 10000); freeswitch::consoleLog ("info","Digit>>".$digit."<<"); if (! $session->ready() ) { freeswitch::consoleLog("info","Going to Exit\n"); last; } if (defined $digit and $digit ne "" ) { freeswitch::consoleLog("info","DTMF received: $digit\n"); if ($digit eq '#') { return; } } else { freeswitch::consoleLog("info","Timeout\n"); $session->hangup(); } } 1; If my understanding is right then, I believe there is something wrong with channel_variable. Kindly help me to resolve this. -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/bd57341f/attachment.html From babak.freeswitch at gmail.com Tue Jun 15 00:16:21 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Tue, 15 Jun 2010 11:46:21 +0430 Subject: [Freeswitch-users] extension mobility using freeswitch Message-ID: Hi Is it possible to implement something like cisco ccm extension mobility using freeswitch and cisco xml objects and services for cisco ip phones? thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/8d590dd3/attachment.html From math.parent at gmail.com Tue Jun 15 00:49:06 2010 From: math.parent at gmail.com (Mathieu Parent) Date: Tue, 15 Jun 2010 09:49:06 +0200 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: References: Message-ID: Can you describe in-depth what it means? It may be possible to implement this within mod_skinny, but we need network captures of CCM behavior and description of features. You can create a jira with all those informations. Mathieu Parent On Tue, Jun 15, 2010 at 9:16 AM, babak yakhchali wrote: > Hi > Is it possible to implement something like cisco ccm extension mobility > using freeswitch and cisco xml objects and services for cisco ip phones? > thanx > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 dcolombo at voismart.it Tue Jun 15 01:46:00 2010 From: dcolombo at voismart.it (Davide Colombo) Date: Tue, 15 Jun 2010 10:46:00 +0200 (CEST) Subject: [Freeswitch-users] Question about execute_on_answer In-Reply-To: <261229835.2605.1276591347125.JavaMail.root@mx.voismart.com> Message-ID: <1788278806.2614.1276591560710.JavaMail.root@mx.voismart.com> Hi, using group_confirm_file and group_confirm_key i have only one choice, connect called party with calling party (1 dtmf maps with group_confirm_key). What i'd like to do when called party answers, it's: press 1 to connect with calling party, press 2 to hangup calling party, press 3 move calling party to voicemail. Additionally, making parellel dial in bridge, called phones stop to ring only when one of them digits the group_confirm_key and not (my purpose) when one of them answers. These are the reasons to transfer called channel to an ivr context using "execute_on_answer" variable. Best Regards ----- Messaggio originale ----- Da: "Michael Collins" A: freeswitch-users at lists.freeswitch.org Inviato: Marted?, 15 giugno 2010 0:37:18 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer Aha! You are trying to do some sort of answer verification, correct? You do know that FS already has this feature, correct? Check it out: http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables Make sure that you aren't duplicating functionality before continuing. -MC On Mon, Jun 14, 2010 at 3:12 PM, Davide Colombo < dcolombo at voismart.it > wrote: Hi, when 1019 answers, i transfer its channel (called channel party) to an IVR context using "execute_on_answer". In this context the called party (1019 in this case) can listen to a message where he can choose to connect/hang-up the calling party using dtmf tones. In the meanwhile i need that calling party remains in a ringing state. Best regards ----- Messaggio originale ----- Da: "Michael Collins" < msc at freeswitch.org > A: freeswitch-users at lists.freeswitch.org Inviato: Luned?, 14 giugno 2010 23:46:42 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer I'm not sure I follow what you're trying to accomplish... You call from x1000 to x1019... and what exactly do you want to have happen when 1019 answers? -MC On Mon, Jun 14, 2010 at 8:22 AM, Davide Colombo < dcolombo at voismart.it > wrote: Hi all, i'm trying to use "execute_on_answer" channel variable with a transfer before a bridge: When i call 1019 from 1000 and 1019 answers, i have this situation: 1019 is transfered to myivrpresentation but in console i can see this error 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 sofia/internal/ 1000 at fs2-devel.voismart.net Restore previous codec PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] while 1000 goes to next action after bridge in the same extension. Is it possible to transfer called channel (when called party answers) to a particular extension and leave caller channel in ringing state? Best Regards _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 From steveayre at gmail.com Tue Jun 15 02:06:37 2010 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 15 Jun 2010 10:06:37 +0100 Subject: [Freeswitch-users] Question about execute_on_answer In-Reply-To: <1788278806.2614.1276591560710.JavaMail.root@mx.voismart.com> References: <261229835.2605.1276591347125.JavaMail.root@mx.voismart.com> <1788278806.2614.1276591560710.JavaMail.root@mx.voismart.com> Message-ID: Read the "exec in answer confirm" section of http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation I think you would be able to do what you're asking within a script that's executed this way, playing the file and reading DTMF then doing logic based on the pressed key from within the script. Any scripting language should work - so lua etc as well as javascript. -Steve On 15 June 2010 09:46, Davide Colombo wrote: > Hi, > > using group_confirm_file and group_confirm_key i have only one choice, > connect called party with calling party (1 dtmf maps with > group_confirm_key). > What i'd like to do when called party answers, it's: press 1 to connect > with calling party, press 2 to hangup calling party, press 3 move calling > party to voicemail. > Additionally, making parellel dial in bridge, called phones stop to ring > only when one of them digits the group_confirm_key and not (my purpose) when > one of them answers. > > These are the reasons to transfer called channel to an ivr context using > "execute_on_answer" variable. > > Best Regards > > > > ----- Messaggio originale ----- > Da: "Michael Collins" > A: freeswitch-users at lists.freeswitch.org > Inviato: Marted?, 15 giugno 2010 0:37:18 > Oggetto: Re: [Freeswitch-users] Question about execute_on_answer > > Aha! > > You are trying to do some sort of answer verification, correct? You do > know that FS already has this feature, correct? Check it out: > > http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables > > Make sure that you aren't duplicating functionality before continuing. > -MC > > > On Mon, Jun 14, 2010 at 3:12 PM, Davide Colombo < dcolombo at voismart.it > > wrote: > > > Hi, > when 1019 answers, i transfer its channel (called channel party) to an > IVR context using "execute_on_answer". In this context the called party > (1019 in this case) can listen to a message where he can choose to > connect/hang-up the calling party using dtmf tones. > In the meanwhile i need that calling party remains in a ringing state. > > Best regards > > ----- Messaggio originale ----- > Da: "Michael Collins" < msc at freeswitch.org > > A: freeswitch-users at lists.freeswitch.org > Inviato: Luned?, 14 giugno 2010 23:46:42 > Oggetto: Re: [Freeswitch-users] Question about execute_on_answer > > > > > I'm not sure I follow what you're trying to accomplish... > > You call from x1000 to x1019... and what exactly do you want to have > happen when 1019 answers? > > -MC > > > On Mon, Jun 14, 2010 at 8:22 AM, Davide Colombo < dcolombo at voismart.it > > wrote: > > > Hi all, > > i'm trying to use "execute_on_answer" channel variable with a transfer > before a bridge: > > > > > When i call 1019 from 1000 and 1019 answers, i have this situation: > > 1019 is transfered to myivrpresentation but in console i can see this > error > > 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 > sofia/internal/ 1000 at fs2-devel.voismart.net Restore previous codec > PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 > Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot > create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] > 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > > while 1000 goes to next action after bridge in the same extension. > > Is it possible to transfer called channel (when called party answers) to > a particular extension and leave caller channel in ringing state? > > > Best Regards > > _______________________________________________ FreeSWITCH-users mailing > list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > _______________________________________________ FreeSWITCH-users mailing > list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/0d4147de/attachment.html From fraserredmond at gmail.com Tue Jun 15 04:20:31 2010 From: fraserredmond at gmail.com (Fraser Redmond) Date: Tue, 15 Jun 2010 12:20:31 +0100 Subject: [Freeswitch-users] New provider based on asterisk instructions (xnet/worldxchange in New Zealand) Message-ID: Hi, I'm trying to set up a new provider, based on the instructions they've provided for Asterisk. I'm primarily interested in inbound calls for now. I've set up new providers a couple of times before, but my guesswork isn't working out this time. I'm getting a 404 error. The instructions say: Edit your Sip_additional.conf file and add the following: [VFX] type=peer fromuser=DIDnumber host=pan.wxnz.net insecure=invite,port canreinvite=no nat=yes secret=PASSWORD username=USERID register=DIDnumber:PASSWORD:USERID at pan.wxnz.net/DIDnumber I suspect the problem is to do with the register line, any ideas on how to set that up in FreeSwitch? Or any other suggestions of settings I ought to try changing? Cheers, Fraser -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/a3d63d52/attachment.html From fraserredmond at gmail.com Tue Jun 15 04:28:20 2010 From: fraserredmond at gmail.com (Fraser Redmond) Date: Tue, 15 Jun 2010 12:28:20 +0100 Subject: [Freeswitch-users] New provider based on asterisk instructions (xnet/worldxchange in New Zealand) In-Reply-To: References: Message-ID: Um... never mind... as usual, you spend a few hours working on something, ask for help and the very next thing you try gets it working :-) (Well partly working, but no point anyone else spending time on this for now.) Cheers, Fraser On Tue, Jun 15, 2010 at 12:20 PM, Fraser Redmond wrote: > Hi, > > I'm trying to set up a new provider, based on the instructions they've > provided for Asterisk. I'm primarily interested in inbound calls for now. > > I've set up new providers a couple of times before, but my guesswork isn't > working out this time. > > I'm getting a 404 error. > > The instructions say: > > Edit your Sip_additional.conf file and add the following: > [VFX] > type=peer > fromuser=DIDnumber > host=pan.wxnz.net > insecure=invite,port > canreinvite=no > nat=yes > secret=PASSWORD > username=USERID > > register=DIDnumber:PASSWORD:USERID at pan.wxnz.net/DIDnumber > > > I suspect the problem is to do with the register line, any ideas on how to > set that up in FreeSwitch? Or any other suggestions of settings I ought to > try changing? > > Cheers, > Fraser > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/4020620f/attachment.html From david.ponzone at gmail.com Tue Jun 15 04:35:49 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Tue, 15 Jun 2010 13:35:49 +0200 Subject: [Freeswitch-users] New provider based on asterisk instructions (xnet/worldxchange in New Zealand) In-Reply-To: References: Message-ID: <0F593816-637E-4797-8DC6-72DA9816EDEE@gmail.com> Fraser, in conf/sip_profiles/external/, you would add a foo.xml file including something like this: then in CLI, do: sofia profile external rescan reloadxml David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 15/06/2010 ? 13:20, Fraser Redmond a ?crit : > Hi, > > I'm trying to set up a new provider, based on the instructions > they've provided for Asterisk. I'm primarily interested in inbound > calls for now. > > I've set up new providers a couple of times before, but my guesswork > isn't working out this time. > > I'm getting a 404 error. > > The instructions say: > > Edit your Sip_additional.conf file and add the following: > [VFX] > type=peer > fromuser=DIDnumber > host=pan.wxnz.net > insecure=invite,port > canreinvite=no > nat=yes > secret=PASSWORD > username=USERID > > register=DIDnumber:PASSWORD:USERID at pan.wxnz.net/DIDnumber > > > I suspect the problem is to do with the register line, any ideas on > how to set that up in FreeSwitch? Or any other suggestions of > settings I ought to try changing? > > Cheers, > Fraser > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/ae140836/attachment-0001.html From samu60 at gmail.com Tue Jun 15 05:03:52 2010 From: samu60 at gmail.com (samuel) Date: Tue, 15 Jun 2010 14:03:52 +0200 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries Message-ID: hi all, recently testing T.38 unsuccesfully with freeswitch head version (FreeSWITCH Version 1.0.head (git-) ) on a testing server (2.6.26-2-xen-amd64) remote SDP: v=0 o=root 6039 6040 IN IP4 A.B.C.D s=session c=IN IP4 A.B.C.D t=0 0 m=image 19423 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy freeswitch answer: v=0 o=FreeSWITCH 1276579579 1276579581 IN IP4 W.X.Y.Z s=FreeSWITCH c=IN IP4 W.X.Y.Z t=0 0 m=image 22230 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 some debug: 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Too many retries. Giving up. 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set rx type 0 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set tx type 4 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from state 17 to 3 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: DCN with final frame tag 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: ff 13 fa 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Disconnecting 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set rx type 0 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set tx type 1 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from state 3 to 2 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send complete in phase T30_PHASE_E, state 2 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:301 ============================================================================== 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:313 Fax processing not successful - result (48) Disconnected after permitted retries. 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:318 Remote station id: 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:319 Local station id: SpanDSP Fax Ident 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:320 Pages transferred: 0 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:322 Total fax pages: 0 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:323 Image resolution: 0x0 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:324 Transfer Rate: 14400 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:326 ECM status off I've starting learning about T3X and freswitch so I would like whether someone can point me to any doc or any configuration parameter that can solve the issue or at least debug it. Thanks in advance, Samuel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/ded06891/attachment.html From luixsansan at hotmail.com Tue Jun 15 02:44:41 2010 From: luixsansan at hotmail.com (luixsansan at hotmail.com) Date: Tue, 15 Jun 2010 11:44:41 +0200 Subject: [Freeswitch-users] vars.xml for Spain Message-ID: Hello Reading the vars.xml file I found the following defaults I will assume default_areacode=91 for Madrid although the area code is added to all numbers in Spain and I wonder if it is better to leave this field blank, default_country will be ES but which is the es-ring, bong-ring and sit for Spain? Thank you very much for your help. Luis. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/d07b0418/attachment.html From mike at jerris.com Tue Jun 15 08:08:00 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Jun 2010 11:08:00 -0400 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: References: Message-ID: <5F5B51F8-1310-4CCF-B158-FE45CF7C42EB@jerris.com> what actual version is this? seems that its cut off. Mike On Jun 15, 2010, at 8:03 AM, samuel wrote: > hi all, > > recently testing T.38 unsuccesfully with freeswitch head version (FreeSWITCH Version 1.0.head (git-) ) on a testing server (2.6.26-2-xen-amd64) > > remote SDP: > > v=0 > o=root 6039 6040 IN IP4 A.B.C.D > s=session > c=IN IP4 A.B.C.D > t=0 0 > m=image 19423 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:72 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > > freeswitch answer: > v=0 > o=FreeSWITCH 1276579579 1276579581 IN IP4 W.X.Y.Z > s=FreeSWITCH > c=IN IP4 W.X.Y.Z > t=0 0 > m=image 22230 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > > some debug: > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Too many retries. Giving up. > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set rx type 0 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set tx type 4 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from state 17 to 3 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: DCN with final frame tag > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: ff 13 fa > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Disconnecting > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set rx type 0 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set tx type 1 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from state 3 to 2 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send complete in phase T30_PHASE_E, state 2 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:301 ============================================================================== > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:313 Fax processing not successful - result (48) Disconnected after permitted retries. > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:318 Remote station id: > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:319 Local station id: SpanDSP Fax Ident > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:320 Pages transferred: 0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:322 Total fax pages: 0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:323 Image resolution: 0x0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:324 Transfer Rate: 14400 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:326 ECM status off > > I've starting learning about T3X and freswitch so I would like whether someone can point me to any doc or any configuration parameter that can solve the issue or at least debug it. From anthony.minessale at gmail.com Tue Jun 15 08:12:41 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Jun 2010 10:12:41 -0500 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: References: Message-ID: did you watch it in a pcap? Maybe it's one way audio. On Tue, Jun 15, 2010 at 7:03 AM, samuel wrote: > hi all, > > recently testing T.38 unsuccesfully with freeswitch head version > (FreeSWITCH Version 1.0.head (git-) ) on a testing server > (2.6.26-2-xen-amd64) > > remote SDP: > > v=0 > o=root 6039 6040 IN IP4 A.B.C.D > s=session > c=IN IP4 A.B.C.D > t=0 0 > m=image 19423 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:72 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > > freeswitch answer: > v=0 > o=FreeSWITCH 1276579579 1276579581 IN IP4 W.X.Y.Z > s=FreeSWITCH > c=IN IP4 W.X.Y.Z > t=0 0 > m=image 22230 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > > some debug: > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 T4 > expired in phase T30_PHASE_B_RX, state 17 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Too many > retries. Giving up. > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing > from phase T30_PHASE_B_RX to T30_PHASE_D_TX > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set rx > type 0 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set tx > type 4 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing > from state 17 to 3 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: DCN > with final frame tag > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: ff > 13 fa > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send > complete in phase T30_PHASE_D_TX, state 3 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send > complete in phase T30_PHASE_D_TX, state 3 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Disconnecting > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing > from phase T30_PHASE_D_TX to T30_PHASE_E > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set rx > type 0 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set tx > type 1 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing > from state 3 to 2 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send > complete in phase T30_PHASE_E, state 2 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:301 > ============================================================================== > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:313 Fax processing not > successful - result (48) Disconnected after permitted retries. > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:318 Remote station id: > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:319 Local station id: > SpanDSP Fax Ident > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:320 Pages transferred: > 0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:322 Total fax pages: > 0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:323 Image resolution: > 0x0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:324 Transfer Rate: > 14400 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:326 ECM status > off > > I've starting learning about T3X and freswitch so I would like whether > someone can point me to any doc or any configuration parameter that can > solve the issue or at least debug it. > > Thanks in advance, > > Samuel > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100615/a19ba709/attachment-0001.html From samu60 at gmail.com Tue Jun 15 08:35:10 2010 From: samu60 at gmail.com (samuel) Date: Tue, 15 Jun 2010 17:35:10 +0200 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: <5F5B51F8-1310-4CCF-B158-FE45CF7C42EB@jerris.com> References: <5F5B51F8-1310-4CCF-B158-FE45CF7C42EB@jerris.com> Message-ID: last one...I created the local git version a few hours ago... On 15 June 2010 17:08, Michael Jerris wrote: > what actual version is this? seems that its cut off. > > Mike > > On Jun 15, 2010, at 8:03 AM, samuel wrote: > > > hi all, > > > > recently testing T.38 unsuccesfully with freeswitch head version > (FreeSWITCH Version 1.0.head (git-) ) on a testing server > (2.6.26-2-xen-amd64) > > > > remote SDP: > > > > v=0 > > o=root 6039 6040 IN IP4 A.B.C.D > > s=session > > c=IN IP4 A.B.C.D > > t=0 0 > > m=image 19423 udptl t38 > > a=T38FaxVersion:0 > > a=T38MaxBitRate:9600 > > a=T38FaxFillBitRemoval:0 > > a=T38FaxTranscodingMMR:0 > > a=T38FaxTranscodingJBIG:0 > > a=T38FaxRateManagement:transferredTCF > > a=T38FaxMaxBuffer:72 > > a=T38FaxMaxDatagram:72 > > a=T38FaxUdpEC:t38UDPRedundancy > > > > freeswitch answer: > > v=0 > > o=FreeSWITCH 1276579579 1276579581 IN IP4 W.X.Y.Z > > s=FreeSWITCH > > c=IN IP4 W.X.Y.Z > > t=0 0 > > m=image 22230 udptl t38 > > a=T38FaxVersion:0 > > a=T38MaxBitRate:14400 > > a=T38FaxFillBitRemoval > > a=T38FaxRateManagement:transferredTCF > > a=T38FaxMaxBuffer:2000 > > a=T38FaxMaxDatagram:400 > > a=T38FaxUdpEC:t38UDPRedundancy > > a=T38VendorInfo:0 0 0 > > > > > > some debug: > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 T4 > expired in phase T30_PHASE_B_RX, state 17 > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Too > many retries. Giving up. > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set > rx type 0 > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set > tx type 4 > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Changing from state 17 to 3 > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: > DCN with final frame tag > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: > ff 13 fa > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send > complete in phase T30_PHASE_D_TX, state 3 > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send > complete in phase T30_PHASE_D_TX, state 3 > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Disconnecting > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Changing from phase T30_PHASE_D_TX to T30_PHASE_E > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set > rx type 0 > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set > tx type 1 > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Changing from state 3 to 2 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send > complete in phase T30_PHASE_E, state 2 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:301 > ============================================================================== > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:313 Fax processing > not successful - result (48) Disconnected after permitted retries. > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:318 Remote station > id: > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:319 Local station > id: SpanDSP Fax Ident > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:320 Pages > transferred: 0 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:322 Total fax pages: > 0 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:323 Image > resolution: 0x0 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:324 Transfer Rate: > 14400 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:326 ECM status > off > > > > I've starting learning about T3X and freswitch so I would like whether > someone can point me to any doc or any configuration parameter that can > solve the issue or at least debug it. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/4aba3995/attachment.html From samu60 at gmail.com Tue Jun 15 08:41:38 2010 From: samu60 at gmail.com (samuel) Date: Tue, 15 Jun 2010 17:41:38 +0200 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: References: Message-ID: I got a pcap file and after the T.38 renegotiation, wireshark reports T.38 Malformed packet from the "originator" A.B.C.D. It sends 3 of these packets and then stops. From freeswitch I just see "Unknown RTP version 0" packets going to A.B.C.D I attach a png containing the wireshark malformed packet "capture"..apologies in advanced but it's a few Kb and it's the fastest way not to replace all "sensitive" information (IPs, hostnames,etc...). Hope it passes the list filter.. On 15 June 2010 17:12, Anthony Minessale wrote: > did you watch it in a pcap? > Maybe it's one way audio. > > > > On Tue, Jun 15, 2010 at 7:03 AM, samuel wrote: > >> hi all, >> >> recently testing T.38 unsuccesfully with freeswitch head version >> (FreeSWITCH Version 1.0.head (git-) ) on a testing server >> (2.6.26-2-xen-amd64) >> >> remote SDP: >> >> v=0 >> o=root 6039 6040 IN IP4 A.B.C.D >> s=session >> c=IN IP4 A.B.C.D >> t=0 0 >> m=image 19423 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:72 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> freeswitch answer: >> v=0 >> o=FreeSWITCH 1276579579 1276579581 IN IP4 W.X.Y.Z >> s=FreeSWITCH >> c=IN IP4 W.X.Y.Z >> t=0 0 >> m=image 22230 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> >> some debug: >> 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 T4 >> expired in phase T30_PHASE_B_RX, state 17 >> 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Too >> many retries. Giving up. >> 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 >> Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX >> 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set rx >> type 0 >> 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set tx >> type 4 >> 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 >> Changing from state 17 to 3 >> 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: >> DCN with final frame tag >> 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: ff >> 13 fa >> 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send >> complete in phase T30_PHASE_D_TX, state 3 >> 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send >> complete in phase T30_PHASE_D_TX, state 3 >> 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 >> Disconnecting >> 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 >> Changing from phase T30_PHASE_D_TX to T30_PHASE_E >> 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set rx >> type 0 >> 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set tx >> type 1 >> 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 >> Changing from state 3 to 2 >> 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send >> complete in phase T30_PHASE_E, state 2 >> 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:301 >> ============================================================================== >> 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:313 Fax processing >> not successful - result (48) Disconnected after permitted retries. >> 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:318 Remote station >> id: >> 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:319 Local station >> id: SpanDSP Fax Ident >> 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:320 Pages >> transferred: 0 >> 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:322 Total fax >> pages: 0 >> 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:323 Image >> resolution: 0x0 >> 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:324 Transfer >> Rate: 14400 >> 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:326 ECM >> status off >> >> I've starting learning about T3X and freswitch so I would like whether >> someone can point me to any doc or any configuration parameter that can >> solve the issue or at least debug it. >> >> Thanks in advance, >> >> Samuel >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/6de7e339/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: t38.png Type: image/png Size: 26301 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/6de7e339/attachment-0001.png From david.ponzone at gmail.com Tue Jun 15 08:48:24 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Tue, 15 Jun 2010 17:48:24 +0200 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: References: Message-ID: <045EBE21-A71A-46D1-B3DC-E753B8C246B3@gmail.com> Just in case, do you know what device is on the other end ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 15/06/2010 ? 17:41, samuel a ?crit : > I got a pcap file and after the T.38 renegotiation, wireshark > reports T.38 Malformed packet from the "originator" A.B.C.D. It > sends 3 of these packets and then stops. From freeswitch I just see > "Unknown RTP version 0" packets going to A.B.C.D > > I attach a png containing the wireshark malformed packet > "capture"..apologies in advanced but it's a few Kb and it's the > fastest way not to replace all "sensitive" information (IPs, > hostnames,etc...). Hope it passes the list filter.. > > > On 15 June 2010 17:12, Anthony Minessale > wrote: > did you watch it in a pcap? > Maybe it's one way audio. > > > > On Tue, Jun 15, 2010 at 7:03 AM, samuel wrote: > hi all, > > recently testing T.38 unsuccesfully with freeswitch head version > (FreeSWITCH Version 1.0.head (git-) ) on a testing server (2.6.26-2- > xen-amd64) > > remote SDP: > > v=0 > o=root 6039 6040 IN IP4 A.B.C.D > s=session > c=IN IP4 A.B.C.D > t=0 0 > m=image 19423 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:72 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > > freeswitch answer: > v=0 > o=FreeSWITCH 1276579579 1276579581 IN IP4 W.X.Y.Z > s=FreeSWITCH > c=IN IP4 W.X.Y.Z > t=0 0 > m=image 22230 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > > some debug: > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > T4 expired in phase T30_PHASE_B_RX, state 17 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Too many retries. Giving up. > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T > Set rx type 0 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T > Set tx type 4 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Changing from state 17 to 3 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Tx: DCN with final frame tag > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Tx: ff 13 fa > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Send complete in phase T30_PHASE_D_TX, state 3 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Send complete in phase T30_PHASE_D_TX, state 3 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Disconnecting > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Changing from phase T30_PHASE_D_TX to T30_PHASE_E > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T > Set rx type 0 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T > Set tx type 1 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Changing from state 3 to 2 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 > Send complete in phase T30_PHASE_E, state 2 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:301 > = > = > = > = > = > = > = > = > ====================================================================== > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:313 Fax > processing not successful - result (48) Disconnected after permitted > retries. > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:318 Remote > station id: > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:319 Local > station id: SpanDSP Fax Ident > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:320 Pages > transferred: 0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:322 Total fax > pages: 0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:323 Image > resolution: 0x0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:324 Transfer > Rate: 14400 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:326 ECM > status off > > I've starting learning about T3X and freswitch so I would like > whether someone can point me to any doc or any configuration > parameter that can solve the issue or at least debug it. > > Thanks in advance, > > Samuel > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100615/6969a973/attachment.html From freeswitch-list at puzzled.xs4all.nl Tue Jun 15 09:26:22 2010 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Tue, 15 Jun 2010 18:26:22 +0200 Subject: [Freeswitch-users] vars.xml for Spain In-Reply-To: References: Message-ID: <4C17A9AE.3030604@puzzled.xs4all.nl> On 06/15/2010 11:44 AM, luixsansan at hotmail.com wrote: > Hello > Reading the vars.xml file I found the following defaults > > > > > > data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/> > > > > > 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)"/> > I will assume default_areacode=91 for Madrid although the area code is > added to all numbers in Spain and I wonder if it is better to leave this > field blank, default_country will be ES but which is the es-ring, > bong-ring and sit for Spain? There is a link on the wiki to an ITU document that describes the various tones: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf Regards, Patrick From sean at obscuradigital.com Tue Jun 15 10:03:53 2010 From: sean at obscuradigital.com (Sean Holt) Date: Tue, 15 Jun 2010 10:03:53 -0700 Subject: [Freeswitch-users] 4 second delay In-Reply-To: Message-ID: Hey MC, would you be willing to look over my tcpdump to help determine where the failure is happening? Sean On 6/14/10 12:32 PM, "Michael Collins" wrote: > Hop on the console, turn on siptrace and watch the call flow for clues. You > might need to do a tcpdump capturing both signaling and media and then analyze > in Wireshark to see what exactly is happening. > -MC > > On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt wrote: >> Hello list, >> >> I?ve been dealing with a particular issue with in-coming calls. ? >> Leg A calls into the office, then Leg B (endpoint) picks up call but hears >> nothing on other side. ?Wait 4 sec call completes and Leg B can hear the >> other person. ? >> >> I have Centos 5.4 >> Latest git build >> Polycom phones >> >> Calling out is not a problem. >> >> Not sure how to troubleshoot this issue or maybe there?s a delay setting in >> the sip profile that waits for the channel to complete. >> >> Thanks for the help >> Sean >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100615/7439cd88/attachment-0001.html From steveu at coppice.org Tue Jun 15 10:17:48 2010 From: steveu at coppice.org (Steve Underwood) Date: Wed, 16 Jun 2010 01:17:48 +0800 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: References: Message-ID: <4C17B5BC.5050608@coppice.org> Hi, That png looks like an RTP packet being interpreted as if it were UDPTL. This is quite normal during switchover, as the two ends never switch at the same moment. If you see nothing arriving after those three packets, it looks like no T.38 packets arrive at all. Steve On 06/15/2010 11:41 PM, samuel wrote: > I got a pcap file and after the T.38 renegotiation, wireshark reports > T.38 Malformed packet from the "originator" A.B.C.D. It sends 3 of > these packets and then stops. From freeswitch I just see "Unknown RTP > version 0" packets going to A.B.C.D > > I attach a png containing the wireshark malformed packet > "capture"..apologies in advanced but it's a few Kb and it's the > fastest way not to replace all "sensitive" information (IPs, > hostnames,etc...). Hope it passes the list filter.. > > > On 15 June 2010 17:12, Anthony Minessale > wrote: > > did you watch it in a pcap? > Maybe it's one way audio. > > > > On Tue, Jun 15, 2010 at 7:03 AM, samuel > wrote: > > hi all, > > recently testing T.38 unsuccesfully with freeswitch head > version (FreeSWITCH Version 1.0.head (git-) ) on a testing > server (2.6.26-2-xen-amd64) > > remote SDP: > > v=0 > o=root 6039 6040 IN IP4 A.B.C.D > s=session > c=IN IP4 A.B.C.D > t=0 0 > m=image 19423 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:72 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > > freeswitch answer: > v=0 > o=FreeSWITCH 1276579579 1276579581 IN IP4 W.X.Y.Z > s=FreeSWITCH > c=IN IP4 W.X.Y.Z > t=0 0 > m=image 22230 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > > some debug: > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 T4 expired in phase T30_PHASE_B_RX, state 17 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Too many retries. Giving up. > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.38T Set rx type 0 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.38T Set tx type 4 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Changing from state 17 to 3 > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Tx: DCN with final frame tag > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Tx: ff 13 fa > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Send complete in phase T30_PHASE_D_TX, state 3 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Send complete in phase T30_PHASE_D_TX, state 3 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Disconnecting > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.38T Set rx type 0 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.38T Set tx type 1 > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Changing from state 3 to 2 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:271 FLOW > T.30 Send complete in phase T30_PHASE_E, state 2 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:301 > ============================================================================== > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:313 Fax > processing not successful - result (48) Disconnected after > permitted retries. > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:318 > Remote station id: > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:319 Local > station id: SpanDSP Fax Ident > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:320 Pages > transferred: 0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:322 Total > fax pages: 0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:323 Image > resolution: 0x0 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:324 > Transfer Rate: 14400 > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:326 ECM > status off > > I've starting learning about T3X and freswitch so I would like > whether someone can point me to any doc or any configuration > parameter that can solve the issue or at least debug it. > > Thanks in advance, > > Samuel > From anthony.minessale at gmail.com Tue Jun 15 10:23:03 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Jun 2010 12:23:03 -0500 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: leg timeout beats the group confirm timeouts On Tue, Jun 15, 2010 at 12:28 AM, Nagalenoj H. wrote: > Dear friends, > I've tried using the group_confirm_cancel_timeout channel variable. > I've written a testing script to get digits before bridging. But, it doesn't > seem to be working. > > My understanding after reading wiki is, > * When I dial [leg_timeout=10]user/1005, if he answers before > timeout and in the process of giving digits, then the call shouldn't be > disconnected after the leg_timeout secs (10 sec in the example). > > But, When I experiment it, the call is getting disconnected after 10 > seconds and it doesn't bother whether the callee has answered the > call(Started giving digits) or not answered at all. > > I've checked it with nc as follows, > > sendmsg > call-command: execute > execute-app-name: set > execute-app-arg: group_confirm_key=exec > > sendmsg > call-command: execute > execute-app-name: set > execute-app-arg: group_confirm_file=perl /root/confirm.pl > > sendmsg > call-command: execute > execute-app-name: set > execute-app-arg: group_confirm_cancel_timeout=1 > > sendmsg > call-command: execute > execute-app-name: bridge > execute-app-arg: [leg_timeout=10]user/1005 > > And here is the script, > > use freeswitch; > our $session; > my $digit; > > while(1) { > # Wait till response timeout for the first digit. > $digit = $session->getDigits(1, "", 10000); > freeswitch::consoleLog ("info","Digit>>".$digit."<<"); > > if (! $session->ready() ) { > freeswitch::consoleLog("info","Going to Exit\n"); > last; > } > if (defined $digit and $digit ne "" ) { > freeswitch::consoleLog("info","DTMF received: $digit\n"); > if ($digit eq '#') { > return; > } > } > else { > freeswitch::consoleLog("info","Timeout\n"); > $session->hangup(); > } > } > 1; > > If my understanding is right then, I believe there is something wrong with > channel_variable. > > Kindly help me to resolve this. > > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100615/aeefda58/attachment.html From msc at freeswitch.org Tue Jun 15 11:11:26 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Jun 2010 11:11:26 -0700 Subject: [Freeswitch-users] 4 second delay In-Reply-To: References: Message-ID: Sure. Put all this information into pastebin.freeswitch.org: Your topology, including the SIP provider, any routers/firewalls (including make & model), phone make & model Your relevant dialplan Console log of the call, with SIP trace turned on You can also email me the .pcap file, but I don't want to review it until you've got all the above information posted so that we can get a good picture of what's happening... Thanks! -MC On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt wrote: > Hey MC, would you be willing to look over my tcpdump to help determine > where the failure is happening? > > Sean > > > > On 6/14/10 12:32 PM, "Michael Collins" wrote: > > Hop on the console, turn on siptrace and watch the call flow for clues. You > might need to do a tcpdump capturing both signaling and media and then > analyze in Wireshark to see what exactly is happening. > -MC > > On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt > wrote: > > Hello list, > > I?ve been dealing with a particular issue with in-coming calls. > Leg A calls into the office, then Leg B (endpoint) picks up call but hears > nothing on other side. Wait 4 sec call completes and Leg B can hear the > other person. > > I have Centos 5.4 > Latest git build > Polycom phones > > Calling out is not a problem. > > Not sure how to troubleshoot this issue or maybe there?s a delay setting in > the sip profile that waits for the channel to complete. > > Thanks for the help > Sean > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/74b64f28/attachment.html From msc at freeswitch.org Tue Jun 15 11:22:58 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Jun 2010 11:22:58 -0700 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: On Tue, Jun 15, 2010 at 10:23 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > leg timeout beats the group confirm timeouts > > > FYI, I added this nugget to the group_confirm_cancel_timeout chan var wiki page: http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/52eb69ed/attachment.html From codeghar at gmail.com Tue Jun 15 11:37:02 2010 From: codeghar at gmail.com (Code Ghar) Date: Tue, 15 Jun 2010 13:37:02 -0500 Subject: [Freeswitch-users] Your Security Best Practices Message-ID: Recently I have seen many attempts to break in to VoIP switches I work with. So I thought this might be a good time to discuss everyone's security best practices. If you could relate them to how you implemented them in FreeSWITCH, it would give us beginners good ideas on how to follow the principle of "security first". Let me start off the discussion with some practices we use and have seen in carriers we deal with. Since we only do carrier-to-carrier stuff, it is easy to restrict the IPs where we accept traffic from. This is done using either a dedicated firewall (we use Cisco and pfSense) or host-based (since we use Linux, iptables it is). We restrict ingress traffic on port 5060 (SIP) to trusted sources only. Since a lot of carriers are starting to use re-invites and most do not provide ranges of media IPs they will use, we leave open all UDP ports used by our servers for RTP, usually between 10,000 and 50,000. In this situation, how well does FreeSWITCH cope with attacks? For example, if RTP ports are 10,000 to 20,000, does FS recognize invalid RTP packets and knows how to deal with them? Another scenario is user registration of softphones or ATAs, etc. A practice we use in our lab environment (we will soon be putting servers in production) is to use lengthy passwords, at least 15 characters, mixing alpha-numeric characters. We also do not use 4-digit extensions (default in FS) but use 6-digit. It wasn't started as a security measure but turned out that many attackers we saw usually try 4-digit extensions. So we are safer today (maybe not so in the future when attackers wise up). How do you manage your users, extensions, registrations, etc., from a security aspect? In the same context, we recently saw a flood of invalid user registration requests, at close to 60-80 requests per second, for a sustained period of less than a minute. One proprietary solution we use was unable to handle this flood of requests and dropped 70% of its call before the attacks stopped and it was able to recover. Let's say we use top-of-the-line servers to run FS, maybe two quad-cores with 16GB memory, then should it increase the ability of FS to deal with such unwanted floods? I assume FS is configured with default settings. The real question is: does throwing hardware at the problem solve the issue for FS? And even if it does, is there anything we can do at the FS level to allow it to handle even more flooded traffic on its own, without relying on superior hardware? Then come Man in the Middle attacks. So far I haven't seen a real push in the industry to mitigate these. One common practice is to use prefixes to identify traffic from a source. For example, we can use 999912125550000 where 9999 identifies us to the egress gateway and the rest is the number to call. Of course, since signaling is in plain text someone can analyze this data and then start using the same prefix to send traffic to our egress gateway/carrier. They believe the traffic is from us and charge us for the calls. A second option is to use SIPS but I haven't seen it being used at all. Of course, now we are talking about using TCP instead of UDP, bringing its own advantages and disadvantages. A third option I am starting to see if the use of IPSec tunnels for signaling and RTP is exchanged outside of the tunnel. This works well for carrier-to-carrier traffic but cannot be deployed widely cost-effectively if you do home user to service provider traffic. If you have dealt with any one or all of these scenarios, or have come across others, please share your experiences and solutions with all of us. Of course, if you can also advise how to implement your security best practices in FS it would be just wonderful. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/6a1e026f/attachment-0001.html From sean at obscuradigital.com Tue Jun 15 12:19:38 2010 From: sean at obscuradigital.com (Sean Holt) Date: Tue, 15 Jun 2010 12:19:38 -0700 Subject: [Freeswitch-users] 4 second delay In-Reply-To: Message-ID: Should I put the freeswitch log into the pastebin or attach it? On 6/15/10 11:11 AM, "Michael Collins" wrote: > Sure. Put all this information into pastebin.freeswitch.org > : > > Your topology, including the SIP provider, any routers/firewalls (including > make & model), phone make & model > Your relevant dialplan > Console log of the call, with SIP trace turned on > > You can also email me the .pcap file, but I don't want to review it until > you've got all the above information posted so that we can get a good picture > of what's happening... > > Thanks! > -MC > > > On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt wrote: >> Hey MC, would you be willing to look over my tcpdump to help determine where >> the failure is happening? >> >> Sean >> >> >> >> On 6/14/10 12:32 PM, "Michael Collins" > > wrote: >> >>> Hop on the console, turn on siptrace and watch the call flow for clues. You >>> might need to do a tcpdump capturing both signaling and media and then >>> analyze in Wireshark to see what exactly is happening. >>> -MC >>> >>> On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt >> > wrote: >>>> Hello list, >>>> >>>> I?ve been dealing with a particular issue with in-coming calls. ? >>>> Leg A calls into the office, then Leg B (endpoint) picks up call but hears >>>> nothing on other side. ?Wait 4 sec call completes and Leg B can hear the >>>> other person. ? >>>> >>>> I have Centos 5.4 >>>> Latest git build >>>> Polycom phones >>>> >>>> Calling out is not a problem. >>>> >>>> Not sure how to troubleshoot this issue or maybe there?s a delay setting in >>>> the sip profile that waits for the channel to complete. >>>> >>>> Thanks for the help >>>> Sean >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100615/84ae09aa/attachment.html From msc at freeswitch.org Tue Jun 15 12:45:52 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Jun 2010 12:45:52 -0700 Subject: [Freeswitch-users] 4 second delay In-Reply-To: References: Message-ID: Pastebin, please. On Tue, Jun 15, 2010 at 12:19 PM, Sean Holt wrote: > Should I put the freeswitch log into the pastebin or attach it? > > > On 6/15/10 11:11 AM, "Michael Collins" wrote: > > Sure. Put all this information into pastebin.freeswitch.org < > http://pastebin.freeswitch.org> : > > > Your topology, including the SIP provider, any routers/firewalls (including > make & model), phone make & model > Your relevant dialplan > Console log of the call, with SIP trace turned on > > You can also email me the .pcap file, but I don't want to review it until > you've got all the above information posted so that we can get a good > picture of what's happening... > > Thanks! > -MC > > > On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt > wrote: > > Hey MC, would you be willing to look over my tcpdump to help determine > where the failure is happening? > > Sean > > > > On 6/14/10 12:32 PM, "Michael Collins" http://msc at freeswitch.org> > wrote: > > Hop on the console, turn on siptrace and watch the call flow for clues. You > might need to do a tcpdump capturing both signaling and media and then > analyze in Wireshark to see what exactly is happening. > -MC > > On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt http://sean at obscuradigital.com> > wrote: > > Hello list, > > I?ve been dealing with a particular issue with in-coming calls. > Leg A calls into the office, then Leg B (endpoint) picks up call but hears > nothing on other side. Wait 4 sec call completes and Leg B can hear the > other person. > > I have Centos 5.4 > Latest git build > Polycom phones > > Calling out is not a problem. > > Not sure how to troubleshoot this issue or maybe there?s a delay setting in > the sip profile that waits for the channel to complete. > > Thanks for the help > Sean > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > http://FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > http://FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > > > ------------------------------ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100615/ff377c37/attachment.html From kris at kriskinc.com Tue Jun 15 12:56:59 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 15 Jun 2010 15:56:59 -0400 Subject: [Freeswitch-users] Your Security Best Practices Message-ID: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> What carriers are you using that utilize re-INVITEs? -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces at lists.freeswitch.org < freeswitch-users-bounces at lists.freeswitch.org> *To*: freeswitch-users at lists.freeswitch.org < freeswitch-users at lists.freeswitch.org> *Sent*: Tue Jun 15 14:37:02 2010 *Subject*: [Freeswitch-users] Your Security Best Practices Recently I have seen many attempts to break in to VoIP switches I work with. So I thought this might be a good time to discuss everyone's security best practices. If you could relate them to how you implemented them in FreeSWITCH, it would give us beginners good ideas on how to follow the principle of "security first". Let me start off the discussion with some practices we use and have seen in carriers we deal with. Since we only do carrier-to-carrier stuff, it is easy to restrict the IPs where we accept traffic from. This is done using either a dedicated firewall (we use Cisco and pfSense) or host-based (since we use Linux, iptables it is). We restrict ingress traffic on port 5060 (SIP) to trusted sources only. Since a lot of carriers are starting to use re-invites and most do not provide ranges of media IPs they will use, we leave open all UDP ports used by our servers for RTP, usually between 10,000 and 50,000. In this situation, how well does FreeSWITCH cope with attacks? For example, if RTP ports are 10,000 to 20,000, does FS recognize invalid RTP packets and knows how to deal with them? Another scenario is user registration of softphones or ATAs, etc. A practice we use in our lab environment (we will soon be putting servers in production) is to use lengthy passwords, at least 15 characters, mixing alpha-numeric characters. We also do not use 4-digit extensions (default in FS) but use 6-digit. It wasn't started as a security measure but turned out that many attackers we saw usually try 4-digit extensions. So we are safer today (maybe not so in the future when attackers wise up). How do you manage your users, extensions, registrations, etc., from a security aspect? In the same context, we recently saw a flood of invalid user registration requests, at close to 60-80 requests per second, for a sustained period of less than a minute. One proprietary solution we use was unable to handle this flood of requests and dropped 70% of its call before the attacks stopped and it was able to recover. Let's say we use top-of-the-line servers to run FS, maybe two quad-cores with 16GB memory, then should it increase the ability of FS to deal with such unwanted floods? I assume FS is configured with default settings. The real question is: does throwing hardware at the problem solve the issue for FS? And even if it does, is there anything we can do at the FS level to allow it to handle even more flooded traffic on its own, without relying on superior hardware? Then come Man in the Middle attacks. So far I haven't seen a real push in the industry to mitigate these. One common practice is to use prefixes to identify traffic from a source. For example, we can use 999912125550000 where 9999 identifies us to the egress gateway and the rest is the number to call. Of course, since signaling is in plain text someone can analyze this data and then start using the same prefix to send traffic to our egress gateway/carrier. They believe the traffic is from us and charge us for the calls. A second option is to use SIPS but I haven't seen it being used at all. Of course, now we are talking about using TCP instead of UDP, bringing its own advantages and disadvantages. A third option I am starting to see if the use of IPSec tunnels for signaling and RTP is exchanged outside of the tunnel. This works well for carrier-to-carrier traffic but cannot be deployed widely cost-effectively if you do home user to service provider traffic. If you have dealt with any one or all of these scenarios, or have come across others, please share your experiences and solutions with all of us. Of course, if you can also advise how to implement your security best practices in FS it would be just wonderful. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/e32bba13/attachment-0001.html From stephen at stephenjc.com Tue Jun 15 13:01:00 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Tue, 15 Jun 2010 16:01:00 -0400 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: References: Message-ID: I would also be interested to see some more best practices On Jun 15, 2010 2:37 PM, "Code Ghar" wrote: Recently I have seen many attempts to break in to VoIP switches I work with. So I thought this might be a good time to discuss everyone's security best practices. If you could relate them to how you implemented them in FreeSWITCH, it would give us beginners good ideas on how to follow the principle of "security first". Let me start off the discussion with some practices we use and have seen in carriers we deal with. Since we only do carrier-to-carrier stuff, it is easy to restrict the IPs where we accept traffic from. This is done using either a dedicated firewall (we use Cisco and pfSense) or host-based (since we use Linux, iptables it is). We restrict ingress traffic on port 5060 (SIP) to trusted sources only. Since a lot of carriers are starting to use re-invites and most do not provide ranges of media IPs they will use, we leave open all UDP ports used by our servers for RTP, usually between 10,000 and 50,000. In this situation, how well does FreeSWITCH cope with attacks? For example, if RTP ports are 10,000 to 20,000, does FS recognize invalid RTP packets and knows how to deal with them? Another scenario is user registration of softphones or ATAs, etc. A practice we use in our lab environment (we will soon be putting servers in production) is to use lengthy passwords, at least 15 characters, mixing alpha-numeric characters. We also do not use 4-digit extensions (default in FS) but use 6-digit. It wasn't started as a security measure but turned out that many attackers we saw usually try 4-digit extensions. So we are safer today (maybe not so in the future when attackers wise up). How do you manage your users, extensions, registrations, etc., from a security aspect? In the same context, we recently saw a flood of invalid user registration requests, at close to 60-80 requests per second, for a sustained period of less than a minute. One proprietary solution we use was unable to handle this flood of requests and dropped 70% of its call before the attacks stopped and it was able to recover. Let's say we use top-of-the-line servers to run FS, maybe two quad-cores with 16GB memory, then should it increase the ability of FS to deal with such unwanted floods? I assume FS is configured with default settings. The real question is: does throwing hardware at the problem solve the issue for FS? And even if it does, is there anything we can do at the FS level to allow it to handle even more flooded traffic on its own, without relying on superior hardware? Then come Man in the Middle attacks. So far I haven't seen a real push in the industry to mitigate these. One common practice is to use prefixes to identify traffic from a source. For example, we can use 999912125550000 where 9999 identifies us to the egress gateway and the rest is the number to call. Of course, since signaling is in plain text someone can analyze this data and then start using the same prefix to send traffic to our egress gateway/carrier. They believe the traffic is from us and charge us for the calls. A second option is to use SIPS but I haven't seen it being used at all. Of course, now we are talking about using TCP instead of UDP, bringing its own advantages and disadvantages. A third option I am starting to see if the use of IPSec tunnels for signaling and RTP is exchanged outside of the tunnel. This works well for carrier-to-carrier traffic but cannot be deployed widely cost-effectively if you do home user to service provider traffic. If you have dealt with any one or all of these scenarios, or have come across others, please share your experiences and solutions with all of us. Of course, if you can also advise how to implement your security best practices in FS it would be just wonderful. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/df6cf172/attachment.html From sean at obscuradigital.com Tue Jun 15 14:00:49 2010 From: sean at obscuradigital.com (Sean Holt) Date: Tue, 15 Jun 2010 14:00:49 -0700 Subject: [Freeswitch-users] 4 second delay In-Reply-To: Message-ID: Hey MC, http://pastebin.freeswitch.org/13191 Topology: Bandwidth.com---->ISP provider: Towerstream(8mg up/down)-------->Linksys router(WRT54GL)--------->D-link 24 port DES-3052P------->Polcom phones & Freeswitch server Thanks Sean On 6/15/10 11:11 AM, "Michael Collins" wrote: > Sure. Put all this information into pastebin.freeswitch.org > : > > Your topology, including the SIP provider, any routers/firewalls (including > make & model), phone make & model > Your relevant dialplan > Console log of the call, with SIP trace turned on > > You can also email me the .pcap file, but I don't want to review it until > you've got all the above information posted so that we can get a good picture > of what's happening... > > Thanks! > -MC > > > On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt wrote: >> Hey MC, would you be willing to look over my tcpdump to help determine where >> the failure is happening? >> >> Sean >> >> >> >> On 6/14/10 12:32 PM, "Michael Collins" > > wrote: >> >>> Hop on the console, turn on siptrace and watch the call flow for clues. You >>> might need to do a tcpdump capturing both signaling and media and then >>> analyze in Wireshark to see what exactly is happening. >>> -MC >>> >>> On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt >> > wrote: >>>> Hello list, >>>> >>>> I?ve been dealing with a particular issue with in-coming calls. ? >>>> Leg A calls into the office, then Leg B (endpoint) picks up call but hears >>>> nothing on other side. ?Wait 4 sec call completes and Leg B can hear the >>>> other person. ? >>>> >>>> I have Centos 5.4 >>>> Latest git build >>>> Polycom phones >>>> >>>> Calling out is not a problem. >>>> >>>> Not sure how to troubleshoot this issue or maybe there?s a delay setting in >>>> the sip profile that waits for the channel to complete. >>>> >>>> Thanks for the help >>>> Sean >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100615/cdefab64/attachment.html From codeghar at gmail.com Tue Jun 15 14:01:00 2010 From: codeghar at gmail.com (Code Ghar) Date: Tue, 15 Jun 2010 16:01:00 -0500 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> Message-ID: Let me put it this way: of more than 40 carriers I work with, only six have some sort of problem if they see re-invite. Some of the bigger names most people recognize all support re-invite (and some even prefer it): AT&T, Qwest, and Verizon, among others. Let me clarify when I say re-invite. We are basically talking about keeping yourself in the signaling path but once the call is answered to invite your ingress and egress carriers to use each others' media IP instead of your media IP. This way you don't have to deal with media once call is connected. On Tue, Jun 15, 2010 at 2:56 PM, Kristian Kielhofner wrote: > What carriers are you using that utilize re-INVITEs? > > > -- > Kristian Kielhofner > http://blog.krisk.org > > ------------------------------ > *From*: freeswitch-users-bounces at lists.freeswitch.org < > freeswitch-users-bounces at lists.freeswitch.org> > *To*: freeswitch-users at lists.freeswitch.org < > freeswitch-users at lists.freeswitch.org> > *Sent*: Tue Jun 15 14:37:02 2010 > *Subject*: [Freeswitch-users] Your Security Best Practices > > Recently I have seen many attempts to break in to VoIP switches I work > with. So I thought this might be a good time to discuss everyone's security > best practices. If you could relate them to how you implemented them in > FreeSWITCH, it would give us beginners good ideas on how to follow the > principle of "security first". Let me start off the discussion with some > practices we use and have seen in carriers we deal with. > > Since we only do carrier-to-carrier stuff, it is easy to restrict the IPs > where we accept traffic from. This is done using either a dedicated firewall > (we use Cisco and pfSense) or host-based (since we use Linux, iptables it > is). We restrict ingress traffic on port 5060 (SIP) to trusted sources only. > Since a lot of carriers are starting to use re-invites and most do not > provide ranges of media IPs they will use, we leave open all UDP ports used > by our servers for RTP, usually between 10,000 and 50,000. In this > situation, how well does FreeSWITCH cope with attacks? For example, if RTP > ports are 10,000 to 20,000, does FS recognize invalid RTP packets and knows > how to deal with them? > > Another scenario is user registration of softphones or ATAs, etc. A > practice we use in our lab environment (we will soon be putting servers in > production) is to use lengthy passwords, at least 15 characters, mixing > alpha-numeric characters. We also do not use 4-digit extensions (default in > FS) but use 6-digit. It wasn't started as a security measure but turned out > that many attackers we saw usually try 4-digit extensions. So we are safer > today (maybe not so in the future when attackers wise up). How do you manage > your users, extensions, registrations, etc., from a security aspect? > > In the same context, we recently saw a flood of invalid user registration > requests, at close to 60-80 requests per second, for a sustained period of > less than a minute. One proprietary solution we use was unable to handle > this flood of requests and dropped 70% of its call before the attacks > stopped and it was able to recover. Let's say we use top-of-the-line servers > to run FS, maybe two quad-cores with 16GB memory, then should it increase > the ability of FS to deal with such unwanted floods? I assume FS is > configured with default settings. The real question is: does throwing > hardware at the problem solve the issue for FS? And even if it does, is > there anything we can do at the FS level to allow it to handle even more > flooded traffic on its own, without relying on superior hardware? > > Then come Man in the Middle attacks. So far I haven't seen a real push in > the industry to mitigate these. One common practice is to use prefixes to > identify traffic from a source. For example, we can use 999912125550000 > where 9999 identifies us to the egress gateway and the rest is the number to > call. Of course, since signaling is in plain text someone can analyze this > data and then start using the same prefix to send traffic to our egress > gateway/carrier. They believe the traffic is from us and charge us for the > calls. > > A second option is to use SIPS but I haven't seen it being used at all. Of > course, now we are talking about using TCP instead of UDP, bringing its own > advantages and disadvantages. > > A third option I am starting to see if the use of IPSec tunnels for > signaling and RTP is exchanged outside of the tunnel. This works well for > carrier-to-carrier traffic but cannot be deployed widely cost-effectively if > you do home user to service provider traffic. > > If you have dealt with any one or all of these scenarios, or have come > across others, please share your experiences and solutions with all of us. > Of course, if you can also advise how to implement your security best > practices in FS it would be just wonderful. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/a2eb43f3/attachment-0001.html From brian at freeswitch.org Tue Jun 15 14:11:07 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Jun 2010 16:11:07 -0500 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> Message-ID: <47EBBABB-F2AE-4128-854D-FDEC48020C51@freeswitch.org> We have bypass media. /b On Jun 15, 2010, at 4:01 PM, Code Ghar wrote: > Let me put it this way: of more than 40 carriers I work with, only six have some sort of problem if they see re-invite. Some of the bigger names most people recognize all support re-invite (and some even prefer it): AT&T, Qwest, and Verizon, among others. > > Let me clarify when I say re-invite. We are basically talking about keeping yourself in the signaling path but once the call is answered to invite your ingress and egress carriers to use each others' media IP instead of your media IP. This way you don't have to deal with media once call is connected. From brian at freeswitch.org Tue Jun 15 14:11:27 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Jun 2010 16:11:27 -0500 Subject: [Freeswitch-users] 4 second delay In-Reply-To: References: Message-ID: <5DA13577-6C08-4605-B2EE-3D6DBFBF942E@freeswitch.org> what firmware do you have on that polycom? btw how is towerstream for you? /b On Jun 15, 2010, at 4:00 PM, Sean Holt wrote: > Hey MC, > > http://pastebin.freeswitch.org/13191 > > Topology: > Bandwidth.com---->ISP provider: Towerstream(8mg up/down)-------->Linksys router(WRT54GL)--------->D-link 24 port DES-3052P------->Polcom phones & Freeswitch server > > Thanks > Sean From dswardstrom at remotelink.com Tue Jun 15 14:37:43 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Tue, 15 Jun 2010 16:37:43 -0500 (CDT) Subject: [Freeswitch-users] Javascript Database Timestamp and Date() In-Reply-To: <599753203.228.1276637222109.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <1618876680.234.1276637863775.JavaMail.root@srvr12.remotelinkml.com> I am working with Javascript and Freeswitch to emulate some Conferencing capability that the company I am working for has in production using other products. One thing that I need to do is to access a timestamp from a database in JS. In pgAdmin for Postgres this is a "timestamp without time zone". In JS there is a data type called Date. Question: If I read a row of the database which includes this timestamp field, what should I put it into? * Can I put it directly into a Date data type? * Does it have to come out as a string or a number? * If it is a number can I then put it into a Date? * If it is a string, what is the format: Is it: "YYYY-MM-DD HH:MM:SS.ffffff"? I looked in mod_spidermonkey_odbc.c pdbc_next_row() and it appears to call SQLFetch() and return the result. But information on a row contains both a value but it's type, so what will the interpreter do? If anyone knows, let me know, otherwise I will incorporate a way to test this into my code and let everyone know (may also add this to some Wiki). Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom From jmesquita at freeswitch.org Tue Jun 15 14:47:03 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 15 Jun 2010 18:47:03 -0300 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: <47EBBABB-F2AE-4128-854D-FDEC48020C51@freeswitch.org> References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> <47EBBABB-F2AE-4128-854D-FDEC48020C51@freeswitch.org> Message-ID: I believe it is nice to say as well that most small transit carriers see the bypass media as a commercial threat since they will disclose in part of the packet to which IP address they are actually terminating the call to. I've been through this with lots of carriers that do nothing but LCR dips and transit. JM On Tue, Jun 15, 2010 at 6:11 PM, Brian West wrote: > We have bypass media. > > /b > > On Jun 15, 2010, at 4:01 PM, Code Ghar wrote: > > > Let me put it this way: of more than 40 carriers I work with, only six > have some sort of problem if they see re-invite. Some of the bigger names > most people recognize all support re-invite (and some even prefer it): AT&T, > Qwest, and Verizon, among others. > > > > Let me clarify when I say re-invite. We are basically talking about > keeping yourself in the signaling path but once the call is answered to > invite your ingress and egress carriers to use each others' media IP instead > of your media IP. This way you don't have to deal with media once call is > connected. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/83e87ad3/attachment.html From anthony.minessale at gmail.com Tue Jun 15 14:49:50 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Jun 2010 16:49:50 -0500 Subject: [Freeswitch-users] Javascript Database Timestamp and Date() In-Reply-To: <1618876680.234.1276637863775.JavaMail.root@srvr12.remotelinkml.com> References: <599753203.228.1276637222109.JavaMail.root@srvr12.remotelinkml.com> <1618876680.234.1276637863775.JavaMail.root@srvr12.remotelinkml.com> Message-ID: that would go into a string typically by default, it may be possible to select the date field as epoch time depending on the databse type which is a long int that you could feed to time formatting options of your own. On Tue, Jun 15, 2010 at 4:37 PM, David Swardstrom < dswardstrom at remotelink.com> wrote: > I am working with Javascript and Freeswitch to emulate some Conferencing > capability that the company I am working for has in production using > other products. > > One thing that I need to do is to access a timestamp from a database in JS. > In pgAdmin for Postgres this is a "timestamp without time zone". > > In JS there is a data type called Date. > > Question: If I read a row of the database which includes this timestamp > field, what should I put it into? > * Can I put it directly into a Date data type? > * Does it have to come out as a string or a number? > * If it is a number can I then put it into a Date? > * If it is a string, what is the format: > Is it: "YYYY-MM-DD HH:MM:SS.ffffff"? > > I looked in mod_spidermonkey_odbc.c pdbc_next_row() and it appears to > call SQLFetch() and return the result. > But information on a row contains both a value but it's type, > so what will the interpreter do? > > If anyone knows, let me know, otherwise I will incorporate a way > to test this into my code and let everyone know (may also add this > to some Wiki). > > Regards, > David Swardstrom > (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100615/58643fcc/attachment.html From brian at freeswitch.org Tue Jun 15 14:55:44 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Jun 2010 16:55:44 -0500 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> <47EBBABB-F2AE-4128-854D-FDEC48020C51@freeswitch.org> Message-ID: <094747A4-EB1F-429E-8882-686AC3ECF74F@freeswitch.org> This is what is wrong with this industry. You have every Tom, Dick and Harry setting up Asterisk and FreeSWITCH boxes without an ounce of clue what the heck to do. Then you have mom and pop shops that provide really sub par services by routing media all over the globe when not needed. This results in software such as Asterisk or FreeSWITCH getting a bad name when the media goes to hell because TomDickHarry's voip service he runs on his cable modem went out because it rained outside. So to summarize, either do it right or go home. /b On Jun 15, 2010, at 4:47 PM, Jo?o Mesquita wrote: > I believe it is nice to say as well that most small transit carriers see the bypass media as a commercial threat since they will disclose in part of the packet to which IP address they are actually terminating the call to. > > I've been through this with lots of carriers that do nothing but LCR dips and transit. > > JM From anthony.minessale at gmail.com Tue Jun 15 14:59:42 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Jun 2010 16:59:42 -0500 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> <47EBBABB-F2AE-4128-854D-FDEC48020C51@freeswitch.org> Message-ID: We have some parameters to disable some commonly abused features in a carrier env. in the sofia profile set manage-presence to false If you are not providing registration or client transfers disable them completely set disable-transfer to true set disable-register to true And register for ClueCon where someone will probably give a talk on VoIP security! 2010/6/15 Jo?o Mesquita > I believe it is nice to say as well that most small transit carriers see > the bypass media as a commercial threat since they will disclose in part of > the packet to which IP address they are actually terminating the call to. > > I've been through this with lots of carriers that do nothing but LCR dips > and transit. > > JM > > > > On Tue, Jun 15, 2010 at 6:11 PM, Brian West wrote: > >> We have bypass media. >> >> /b >> >> On Jun 15, 2010, at 4:01 PM, Code Ghar wrote: >> >> > Let me put it this way: of more than 40 carriers I work with, only six >> have some sort of problem if they see re-invite. Some of the bigger names >> most people recognize all support re-invite (and some even prefer it): AT&T, >> Qwest, and Verizon, among others. >> > >> > Let me clarify when I say re-invite. We are basically talking about >> keeping yourself in the signaling path but once the call is answered to >> invite your ingress and egress carriers to use each others' media IP instead >> of your media IP. This way you don't have to deal with media once call is >> connected. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100615/1c40b5c9/attachment-0001.html From sean at obscuradigital.com Tue Jun 15 15:17:58 2010 From: sean at obscuradigital.com (Sean Holt) Date: Tue, 15 Jun 2010 15:17:58 -0700 Subject: [Freeswitch-users] 4 second delay In-Reply-To: <5DA13577-6C08-4605-B2EE-3D6DBFBF942E@freeswitch.org> Message-ID: I have firmware 4.2.0 We've been using Towerstream for about 5 years with about 98% up time. It's not bad. I just started using FS over Towerstream so I don't have any true experience in that regard. Sean On 6/15/10 2:11 PM, "Brian West" wrote: > what firmware do you have on that polycom? btw how is towerstream for you? > > /b > > On Jun 15, 2010, at 4:00 PM, Sean Holt wrote: > >> Hey MC, >> >> http://pastebin.freeswitch.org/13191 >> >> Topology: >> Bandwidth.com---->ISP provider: Towerstream(8mg up/down)-------->Linksys >> router(WRT54GL)--------->D-link 24 port DES-3052P------->Polcom phones & >> Freeswitch server >> >> Thanks >> Sean > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Tue Jun 15 16:55:53 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Jun 2010 16:55:53 -0700 Subject: [Freeswitch-users] Special Announcement: Be On The Community Conf Call Tomorrow! Message-ID: Hey everyone! Brian West has an announcement for everyone so be sure to be on the community conference call tomorrow! The details are in the usual spot on the wiki: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_16 We have updates on the FS project plus a followup presentation by DRK on using mod_managed for CDRs and billing, etc. Talk to you tomorrow! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/a68d1451/attachment.html From jan.berger at video24.no Tue Jun 15 17:06:17 2010 From: jan.berger at video24.no (Jan Berger) Date: Wed, 16 Jun 2010 02:06:17 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] Special Announcement: Be On The Community Conf Call Tomorrow! In-Reply-To: References: Message-ID: <663F299A58AC4E9099617A59AB3E5BEC@dell9400> Oh - is FS running on hes iPad? _____ From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 16. juni 2010 01:56 To: freeswitch-users at lists.freeswitch.org; freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] Special Announcement: Be On The Community Conf Call Tomorrow! Hey everyone! Brian West has an announcement for everyone so be sure to be on the community conference call tomorrow! The details are in the usual spot on the wiki: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_16 We have updates on the FS project plus a followup presentation by DRK on using mod_managed for CDRs and billing, etc. Talk to you tomorrow! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100616/65639db3/attachment.html From rupa at rupa.com Tue Jun 15 17:07:52 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 15 Jun 2010 19:07:52 -0500 Subject: [Freeswitch-users] Javascript Database Timestamp and Date() In-Reply-To: <1618876680.234.1276637863775.JavaMail.root@srvr12.remotelinkml.com> References: <599753203.228.1276637222109.JavaMail.root@srvr12.remotelinkml.com> <1618876680.234.1276637863775.JavaMail.root@srvr12.remotelinkml.com> Message-ID: Postgres has lots of date manipulation functions. So, you should be able to use them to get/set the timestamp to a format that is suitable for whatever you are doing in JS. Look at the date/time functions in the postgres manual. The odbc stuff from squirrelmonkey just gives you the textual representation of whatever you ask for. There is no native type binding. If the Date() datatype takes a constructor then pull the date from the database in an appropriate format. For setting date from a JS Date() you'll want to see what format Date() can give you and look at the timestamp casting / construction possibilities. On Tue, Jun 15, 2010 at 4:37 PM, David Swardstrom < dswardstrom at remotelink.com> wrote: > I am working with Javascript and Freeswitch to emulate some Conferencing > capability that the company I am working for has in production using > other products. > > One thing that I need to do is to access a timestamp from a database in JS. > In pgAdmin for Postgres this is a "timestamp without time zone". > > In JS there is a data type called Date. > > Question: If I read a row of the database which includes this timestamp > field, what should I put it into? > * Can I put it directly into a Date data type? > * Does it have to come out as a string or a number? > * If it is a number can I then put it into a Date? > * If it is a string, what is the format: > Is it: "YYYY-MM-DD HH:MM:SS.ffffff"? > > I looked in mod_spidermonkey_odbc.c pdbc_next_row() and it appears to > call SQLFetch() and return the result. > But information on a row contains both a value but it's type, > so what will the interpreter do? > > If anyone knows, let me know, otherwise I will incorporate a way > to test this into my code and let everyone know (may also add this > to some Wiki). > > Regards, > David Swardstrom > (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100615/fec8d846/attachment.html From brian at freeswitch.org Tue Jun 15 17:13:40 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Jun 2010 19:13:40 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Special Announcement: Be On The Community Conf Call Tomorrow! In-Reply-To: <663F299A58AC4E9099617A59AB3E5BEC@dell9400> References: <663F299A58AC4E9099617A59AB3E5BEC@dell9400> Message-ID: Nope.... Keep guessing... its ClueCon Related.... so get your credit cards ready! ;) /b On Jun 15, 2010, at 7:06 PM, Jan Berger wrote: > Oh ? is FS running on hes iPad? > From babak.freeswitch at gmail.com Tue Jun 15 21:49:22 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Wed, 16 Jun 2010 09:19:22 +0430 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: References: Message-ID: it means u can login on an 79xx cisco ip phone and a device profile is loaded for u on the phone. it needs that u use CiscoIPPhone xml objects and an script to communicate with ip phones to load profiles for a user when he/she logs in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100616/84bd50c5/attachment.html From maciej.aniserowicz at gmail.com Tue Jun 15 23:18:22 2010 From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz) Date: Tue, 15 Jun 2010 23:18:22 -0700 (PDT) Subject: [Freeswitch-users] How to continue call even if recording fails? Message-ID: <1276669102712-5185161.post@n2.nabble.com> Hi, I'm recording sessions by sending uuid_record command to FS after calls are bridged. It works fine, but problems start when i'm sending a wrong/nonexistent directory to create the recorded file in. FS shows error (which is expected and fine): [ERR] mod_sndfile.c:195 Error Opening File [c:/FreeSWITCH/reco/rec_c1e3df00-dda2-4320-8900-4853a216f19b.wav] [System error : The system cannot find the path specified.] [ERR] switch_ivr_async.c:997 Error opening c:/FreeSWITCH/reco/rec_c1e3df00-dda2-4320-8900-4853a216f19b.wav After that the recorded call gets hung up. Is there any way to continue the call even if recording fails? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-continue-call-even-if-recording-fails-tp5185161p5185161.html Sent from the freeswitch-users mailing list archive at Nabble.com. From math.parent at gmail.com Wed Jun 16 00:28:28 2010 From: math.parent at gmail.com (Mathieu Parent) Date: Wed, 16 Jun 2010 09:28:28 +0200 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: References: Message-ID: On Wed, Jun 16, 2010 at 6:49 AM, babak yakhchali wrote: > it means u can login on an 79xx cisco ip phone and a device profile is > loaded for u on the phone. it ?needs that u use CiscoIPPhone xml objects and > an script to communicate with ip phones to load profiles for a user when > he/she logs in I knew the basics of this, but to implement it we need at least a network capture (and attached scenario, scripts and xml objects). Can you provide this? Mathieu Parent From luixsansan at hotmail.com Wed Jun 16 01:39:42 2010 From: luixsansan at hotmail.com (luixsansan at hotmail.com) Date: Wed, 16 Jun 2010 10:39:42 +0200 Subject: [Freeswitch-users] vars.xml for Spain In-Reply-To: <4C17A9AE.3030604@puzzled.xs4all.nl> References: <4C17A9AE.3030604@puzzled.xs4all.nl> Message-ID: Thank you for your answer. -------------------------------------------------- From: "Patrick Lists" Sent: Tuesday, June 15, 2010 6:26 PM To: Subject: Re: [Freeswitch-users] vars.xml for Spain > On 06/15/2010 11:44 AM, luixsansan at hotmail.com wrote: >> Hello >> Reading the vars.xml file I found the following defaults >> >> >> >> >> >> > data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/> >> >> >> >> >> > 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)"/> >> I will assume default_areacode=91 for Madrid although the area code is >> added to all numbers in Spain and I wonder if it is better to leave this >> field blank, default_country will be ES but which is the es-ring, >> bong-ring and sit for Spain? > > There is a link on the wiki to an ITU document that describes the > various tones: > > http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf > > Regards, > Patrick > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 babak.freeswitch at gmail.com Wed Jun 16 02:40:00 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Wed, 16 Jun 2010 14:10:00 +0430 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: References: Message-ID: I've seen it on the wiki of skinny as todo list I think fs developers are working on it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100616/1d34b9f7/attachment.html From samu60 at gmail.com Wed Jun 16 04:44:26 2010 From: samu60 at gmail.com (samuel) Date: Wed, 16 Jun 2010 13:44:26 +0200 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: <4C17B5BC.5050608@coppice.org> References: <4C17B5BC.5050608@coppice.org> Message-ID: I'm trying yo know what the remote end point brand/model/firmware/??? is but I could not get this information yet. The other point says that after the T.38 SDP negotiation, there must be a "negotiation" (after the ACK of the T38 renegotiation there must be an exchange of Send preamble; DIS (and later DCS) that never happens...) and freeswitch does not start it and that is why the transmission results in a call drop after 25 seconds. Thank you very much for your answers so far, Samuel On 15 June 2010 19:17, Steve Underwood wrote: > Hi, > > That png looks like an RTP packet being interpreted as if it were UDPTL. > This is quite normal during switchover, as the two ends never switch at > the same moment. If you see nothing arriving after those three packets, > it looks like no T.38 packets arrive at all. > > Steve > > > On 06/15/2010 11:41 PM, samuel wrote: > > I got a pcap file and after the T.38 renegotiation, wireshark reports > > T.38 Malformed packet from the "originator" A.B.C.D. It sends 3 of > > these packets and then stops. From freeswitch I just see "Unknown RTP > > version 0" packets going to A.B.C.D > > > > I attach a png containing the wireshark malformed packet > > "capture"..apologies in advanced but it's a few Kb and it's the > > fastest way not to replace all "sensitive" information (IPs, > > hostnames,etc...). Hope it passes the list filter.. > > > > > > On 15 June 2010 17:12, Anthony Minessale > > wrote: > > > > did you watch it in a pcap? > > Maybe it's one way audio. > > > > > > > > On Tue, Jun 15, 2010 at 7:03 AM, samuel > > wrote: > > > > hi all, > > > > recently testing T.38 unsuccesfully with freeswitch head > > version (FreeSWITCH Version 1.0.head (git-) ) on a testing > > server (2.6.26-2-xen-amd64) > > > > remote SDP: > > > > v=0 > > o=root 6039 6040 IN IP4 A.B.C.D > > s=session > > c=IN IP4 A.B.C.D > > t=0 0 > > m=image 19423 udptl t38 > > a=T38FaxVersion:0 > > a=T38MaxBitRate:9600 > > a=T38FaxFillBitRemoval:0 > > a=T38FaxTranscodingMMR:0 > > a=T38FaxTranscodingJBIG:0 > > a=T38FaxRateManagement:transferredTCF > > a=T38FaxMaxBuffer:72 > > a=T38FaxMaxDatagram:72 > > a=T38FaxUdpEC:t38UDPRedundancy > > > > freeswitch answer: > > v=0 > > o=FreeSWITCH 1276579579 1276579581 IN IP4 W.X.Y.Z > > s=FreeSWITCH > > c=IN IP4 W.X.Y.Z > > t=0 0 > > m=image 22230 udptl t38 > > a=T38FaxVersion:0 > > a=T38MaxBitRate:14400 > > a=T38FaxFillBitRemoval > > a=T38FaxRateManagement:transferredTCF > > a=T38FaxMaxBuffer:2000 > > a=T38FaxMaxDatagram:400 > > a=T38FaxUdpEC:t38UDPRedundancy > > a=T38VendorInfo:0 0 0 > > > > > > some debug: > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 T4 expired in phase T30_PHASE_B_RX, state 17 > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Too many retries. Giving up. > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.38T Set rx type 0 > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.38T Set tx type 4 > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Changing from state 17 to 3 > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Tx: DCN with final frame tag > > 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Tx: ff 13 fa > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Send complete in phase T30_PHASE_D_TX, state 3 > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Send complete in phase T30_PHASE_D_TX, state 3 > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Disconnecting > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.38T Set rx type 0 > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.38T Set tx type 1 > > 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Changing from state 3 to 2 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:271 FLOW > > T.30 Send complete in phase T30_PHASE_E, state 2 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:301 > > > ============================================================================== > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:313 Fax > > processing not successful - result (48) Disconnected after > > permitted retries. > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:318 > > Remote station id: > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:319 Local > > station id: SpanDSP Fax Ident > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:320 Pages > > transferred: 0 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:322 Total > > fax pages: 0 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:323 Image > > resolution: 0x0 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:324 > > Transfer Rate: 14400 > > 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:326 ECM > > status off > > > > I've starting learning about T3X and freswitch so I would like > > whether someone can point me to any doc or any configuration > > parameter that can solve the issue or at least debug it. > > > > Thanks in advance, > > > > Samuel > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100616/35d81225/attachment.html From steveu at coppice.org Wed Jun 16 05:08:23 2010 From: steveu at coppice.org (Steve Underwood) Date: Wed, 16 Jun 2010 20:08:23 +0800 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: References: <4C17B5BC.5050608@coppice.org> Message-ID: <4C18BEB7.5070702@coppice.org> On 06/16/2010 07:44 PM, samuel wrote: > I'm trying yo know what the remote end point brand/model/firmware/??? > is but I could not get this information yet. > The other point says that after the T.38 SDP negotiation, there must > be a "negotiation" (after the ACK of the T38 renegotiation there must > be an exchange of Send preamble; DIS (and later DCS) that never > happens...) and freeswitch does not start it and that is why the > transmission results in a call drop after 25 seconds. > > > Thank you very much for your answers so far, > Samuel So far you have kept that much secret. What you supplied was only the very last stages of a call log, where Freeswitch seems to be output the messages it should, but gets nothing in return. Steve From math.parent at gmail.com Wed Jun 16 07:01:51 2010 From: math.parent at gmail.com (Mathieu Parent) Date: Wed, 16 Jun 2010 16:01:51 +0200 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: References: Message-ID: On Wed, Jun 16, 2010 at 11:40 AM, babak yakhchali wrote: > I've seen it on the wiki of skinny as todo list I think fs developers are > working on it The TODO list doesn't mean that we are working on it. When I say "we", I can say "me" as I am (as of now) the only dev for mod_skinny. So: no, we are _not_ working on it. "WIP" (Work in progress) means we are working on something. Currently, this is MWI (I will probably commit tonight). I knew the basics of extension mobility, but to implement it we need at least a network capture (and attached scenario, scripts and xml objects). Can you provide this? Mathieu From brian at freeswitch.org Wed Jun 16 07:07:18 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Jun 2010 09:07:18 -0500 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: References: Message-ID: Just to clarify the person (Mathieu Parent) you have been exchanging emails with IS the developer of mod_skinny, I would recommend you collect the data he requested if you want the feature supported. /b On Jun 16, 2010, at 4:40 AM, babak yakhchali wrote: > I've seen it on the wiki of skinny as todo list I think fs developers are working on it _______________________________________________ From samu60 at gmail.com Wed Jun 16 07:15:10 2010 From: samu60 at gmail.com (samuel) Date: Wed, 16 Jun 2010 16:15:10 +0200 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: <4C18BEB7.5070702@coppice.org> References: <4C17B5BC.5050608@coppice.org> <4C18BEB7.5070702@coppice.org> Message-ID: i've provided all the information i've got and what the other point has reported. What extra information do you need? thanks a lot, Samuel. On 16 June 2010 14:08, Steve Underwood wrote: > On 06/16/2010 07:44 PM, samuel wrote: > > I'm trying yo know what the remote end point brand/model/firmware/??? > > is but I could not get this information yet. > > The other point says that after the T.38 SDP negotiation, there must > > be a "negotiation" (after the ACK of the T38 renegotiation there must > > be an exchange of Send preamble; DIS (and later DCS) that never > > happens...) and freeswitch does not start it and that is why the > > transmission results in a call drop after 25 seconds. > > > > > > Thank you very much for your answers so far, > > Samuel > > So far you have kept that much secret. What you supplied was only the > very last stages of a call log, where Freeswitch seems to be output the > messages it should, but gets nothing in return. > > Steve > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100616/71422fec/attachment.html From anthony.minessale at gmail.com Wed Jun 16 09:30:27 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Jun 2010 11:30:27 -0500 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: References: <4C17B5BC.5050608@coppice.org> <4C18BEB7.5070702@coppice.org> Message-ID: The entire log of the call with the debug all the way up including sip traces would be a good start sofia profile internal siptrace on console loglevel debug edit fax.conf.xml and set verbose to true and restart get a pcap of the whole thing. open an issue on jira and attach everything and report the bug ID here. On Wed, Jun 16, 2010 at 9:15 AM, samuel wrote: > i've provided all the information i've got and what the other point has > reported. > > What extra information do you need? > > thanks a lot, > Samuel. > > > On 16 June 2010 14:08, Steve Underwood wrote: > >> On 06/16/2010 07:44 PM, samuel wrote: >> > I'm trying yo know what the remote end point brand/model/firmware/??? >> > is but I could not get this information yet. >> > The other point says that after the T.38 SDP negotiation, there must >> > be a "negotiation" (after the ACK of the T38 renegotiation there must >> > be an exchange of Send preamble; DIS (and later DCS) that never >> > happens...) and freeswitch does not start it and that is why the >> > transmission results in a call drop after 25 seconds. >> > >> > >> > Thank you very much for your answers so far, >> > Samuel >> >> So far you have kept that much secret. What you supplied was only the >> very last stages of a call log, where Freeswitch seems to be output the >> messages it should, but gets nothing in return. >> >> Steve >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100616/b9c31ab2/attachment.html From testeador01 at gmail.com Wed Jun 16 10:06:53 2010 From: testeador01 at gmail.com (Milena) Date: Wed, 16 Jun 2010 12:06:53 -0500 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: References: Message-ID: babak, here is an alternative: If you handle the directory with curl / xml_curl, you can control a variable in the database when you access *55 *66 as sugested (those would be the extensions on the external profile), and then only serve the directory entries for those users who have the variable set to whatever it would be when authenticated by pin; that way your unregistered phones wouldn't match a directory entry and wouldn't register, but then you'll have to deal with the logs being spammed with "you have to create a domain named xxx.." stuff that appears on the logs when users can't register. 2010/6/16 Brian West > Just to clarify the person (Mathieu Parent) you have been exchanging emails > with IS the developer of mod_skinny, I would recommend you collect the data > he requested if you want the feature supported. > > /b > > On Jun 16, 2010, at 4:40 AM, babak yakhchali wrote: > > > I've seen it on the wiki of skinny as todo list I think fs developers are > working on it _______________________________________________ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100616/b8a723cd/attachment.html From testeador01 at gmail.com Wed Jun 16 10:07:56 2010 From: testeador01 at gmail.com (Milena) Date: Wed, 16 Jun 2010 12:07:56 -0500 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: References: Message-ID: wrong thread, this goes for "help about a system with login logout", babak posts a lot -.- 2010/6/16 Milena > babak, here is an alternative: > > If you handle the directory with curl / xml_curl, you can control a > variable in the database when you access *55 *66 as sugested (those would be > the extensions on the external profile), and then only serve the directory > entries for those users who have the variable set to whatever it would be > when authenticated by pin; > > that way your unregistered phones wouldn't match a directory entry and > wouldn't register, but then you'll have to deal with the logs being spammed > with "you have to create a domain named xxx.." stuff that appears on the > logs when users can't register. > > > 2010/6/16 Brian West > > Just to clarify the person (Mathieu Parent) you have been exchanging emails >> with IS the developer of mod_skinny, I would recommend you collect the data >> he requested if you want the feature supported. >> >> /b >> >> On Jun 16, 2010, at 4:40 AM, babak yakhchali wrote: >> >> > I've seen it on the wiki of skinny as todo list I think fs developers >> are working on it _______________________________________________ >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100616/cbac9328/attachment.html From mike at jerris.com Wed Jun 16 10:13:10 2010 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Jun 2010 13:13:10 -0400 Subject: [Freeswitch-users] How to continue call even if recording fails? In-Reply-To: <1276669102712-5185161.post@n2.nabble.com> References: <1276669102712-5185161.post@n2.nabble.com> Message-ID: create the directory and then its not an issue? On Jun 16, 2010, at 2:18 AM, Maciej Aniserowicz wrote: > > Hi, > I'm recording sessions by sending uuid_record command to FS after calls are > bridged. It works fine, but problems start when i'm sending a > wrong/nonexistent directory to create the recorded file in. FS shows error > (which is expected and fine): > > [ERR] mod_sndfile.c:195 Error Opening File > [c:/FreeSWITCH/reco/rec_c1e3df00-dda2-4320-8900-4853a216f19b.wav] [System > error : The system cannot find the path specified.] > [ERR] switch_ivr_async.c:997 Error opening > c:/FreeSWITCH/reco/rec_c1e3df00-dda2-4320-8900-4853a216f19b.wav > > After that the recorded call gets hung up. Is there any way to continue the > call even if recording fails? From arnuld.mizong at gmail.com Wed Jun 16 03:03:59 2010 From: arnuld.mizong at gmail.com (arnuld uttre) Date: Wed, 16 Jun 2010 15:33:59 +0530 Subject: [Freeswitch-users] Call disconnect response from FreeSWITCH Message-ID: I just started using FS (FreeSWITCH). I am connecting to it through telnet as client and giving it originate command to make calls to some SIP phone. Its placing calls without any problems but FS does not send any info when a call completes so I am little confused on how to know that a call is completed. A typical session is shown down here. FS tells immediately that the call is picked up by the person on other side but no information on when a call is completed. Any idea on how to know that a call is completed so that I can free up my malloc()ed memory. [arnuld at dune ~]$ telnet 192.168.0.222 8021 Trying 192.168.0.222... Connected to cobra (192.168.0.222). Escape character is '^]'. Content-Type: auth/request auth **** Content-Type: command/reply Reply-Text: +OK accepted api originate sofia/internal/105 at 192.168.0.228:5060 1111 Content-Type: api/response Content-Length: 41 +OK 19d962e3-9fa1-4f26-ba8c-0b9ac6afadc1 exit Content-Type: command/reply Reply-Text: +OK bye Content-Type: text/disconnect-notice Content-Length: 67 Disconnected, goodbye. See you at ClueCon! http://www.cluecon.com/ Connection closed by foreign host. [arnuld at dune ~]$ -- http://uttre.wordpress.com/2008/05/14/the-lost-love-of-mine/ From maciej.aniserowicz at gmail.com Wed Jun 16 10:28:28 2010 From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz) Date: Wed, 16 Jun 2010 19:28:28 +0200 Subject: [Freeswitch-users] How to continue call even if recording fails? In-Reply-To: References: <1276669102712-5185161.post@n2.nabble.com> Message-ID: <4C1909BC.5040105@gmail.com> Well yes, then it would work. But the recordings directory is configurable setting in my application, which runs on different machine than FreeSWITCH, so I cannot create the folder or check if it exists from the app-level. Is there no other way to do this? MA On 2010-06-16 19:13, Michael Jerris wrote: > create the directory and then its not an issue? > > > On Jun 16, 2010, at 2:18 AM, Maciej Aniserowicz wrote: > > >> Hi, >> I'm recording sessions by sending uuid_record command to FS after calls are >> bridged. It works fine, but problems start when i'm sending a >> wrong/nonexistent directory to create the recorded file in. FS shows error >> (which is expected and fine): >> >> [ERR] mod_sndfile.c:195 Error Opening File >> [c:/FreeSWITCH/reco/rec_c1e3df00-dda2-4320-8900-4853a216f19b.wav] [System >> error : The system cannot find the path specified.] >> [ERR] switch_ivr_async.c:997 Error opening >> c:/FreeSWITCH/reco/rec_c1e3df00-dda2-4320-8900-4853a216f19b.wav >> >> After that the recorded call gets hung up. Is there any way to continue the >> call even if recording fails? >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brian at freeswitch.org Wed Jun 16 10:32:52 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Jun 2010 12:32:52 -0500 Subject: [Freeswitch-users] How to continue call even if recording fails? In-Reply-To: <4C1909BC.5040105@gmail.com> References: <1276669102712-5185161.post@n2.nabble.com> <4C1909BC.5040105@gmail.com> Message-ID: <5C83064A-4A68-44A8-8776-6B99720F4772@freeswitch.org> use the system app to mkdir -p :) /b On Jun 16, 2010, at 12:28 PM, Maciej Aniserowicz wrote: > Well yes, then it would work. But the recordings directory is > configurable setting in my application, which runs on different machine > than FreeSWITCH, so I cannot create the folder or check if it exists > from the app-level. > > Is there no other way to do this? > > MA From brian at freeswitch.org Wed Jun 16 10:33:42 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Jun 2010 12:33:42 -0500 Subject: [Freeswitch-users] How to continue call even if recording fails? In-Reply-To: <4C1909BC.5040105@gmail.com> References: <1276669102712-5185161.post@n2.nabble.com> <4C1909BC.5040105@gmail.com> Message-ID: <9989BE22-A48B-460D-92A2-F0DCB44D2572@freeswitch.org> Just to clarify the reason the call fails is for consistency... Your app is broken if it doesn't make sure the directory is there.... What if this was a case where the phone call could save a client from a HUGE lawsuit and your application failed to actually do anything? See the reasoning? /b On Jun 16, 2010, at 12:28 PM, Maciej Aniserowicz wrote: > Well yes, then it would work. But the recordings directory is > configurable setting in my application, which runs on different machine > than FreeSWITCH, so I cannot create the folder or check if it exists > from the app-level. > > Is there no other way to do this? > > MA From andrew at hijacked.us Wed Jun 16 14:20:06 2010 From: andrew at hijacked.us (Andrew Thompson) Date: Wed, 16 Jun 2010 17:20:06 -0400 Subject: [Freeswitch-users] How to continue call even if recording fails? In-Reply-To: <9989BE22-A48B-460D-92A2-F0DCB44D2572@freeswitch.org> References: <1276669102712-5185161.post@n2.nabble.com> <4C1909BC.5040105@gmail.com> <9989BE22-A48B-460D-92A2-F0DCB44D2572@freeswitch.org> Message-ID: <20100616212006.GD11308@hijacked.us> On Wed, Jun 16, 2010 at 12:33:42PM -0500, Brian West wrote: > Just to clarify the reason the call fails is for consistency... Your app is broken if it doesn't make sure the directory is there.... What if this was a case where the phone call could save a client from a HUGE lawsuit and your application failed to actually do anything? See the reasoning? > Conversely, what if you'd merely like to make a 'best effort' attempt to record the call, because the call is making you lots of money or AVOID this fabled huge lawsuit. Sometimes drives fill up, disks get unmounted, etc and no dialplan magic is going to save you. Personally I hate it when FS drops a call because recording failed, or I tried to uuid_broadcast a nonexistant file, or whatever. Andrew From macedoslm at gmail.com Wed Jun 16 18:50:48 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Wed, 16 Jun 2010 22:50:48 -0300 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: <4C13CAEC.3020005@ewetel.de> References: <4C12D0F7.5000300@ewetel.de> <4C13CAEC.3020005@ewetel.de> Message-ID: Hi Helmut, I got the same error: "No matching function for overloaded 'new_ESLconnection' in /usr/share/php/ESL.php" Have you solved the problem? Thanks, -- Samuel Macedo On 12 June 2010 14:59, Helmut Kuper wrote: > Hi Michael, > > ESL is built and installed, phpmod is built and installed. > > inbound works, test scripts works > > It is just the problem of passing an existing socket to new ESLconnection. > A socket resource is denied, an integer is accepted but is rejected with > "bad file descriptor" > > > > Am 12.06.2010 07:06, schrieb Michael Collins: > > Make sure the ESL is properly built and that the PHP mod is built and > installed. If you did a "make current" recently the you'll need to rebuild > your ESL stuff. > -MC > > On Fri, Jun 11, 2010 at 5:12 PM, Helmut Kuper wrote: > >> Hello, >> >> >> I try to setup a php daemon which uses ESL. >> >> I run the sample php scripts successfully (inbound). Now I want to have >> it outbound to my php daemon socket. >> >> The forked child process which has to interact with incoming tcp >> connection from FS is started successfully. So in this state I have the >> client socket which I have to pass now somehow to ESLconnection I guess. >> >> FS ruby wiki gives this as an example: >> >> @con = ESL::ESLconnection.new(client_socket.fileno) >> >> >> In php I try this: >> >> $con = new ESLconnection($csock); >> >> As a result I got this error: >> >> PHP Fatal error: No matching function for overloaded >> 'new_ESLconnection' in /usr/share/pear/ESL.php on line 117 >> >> >> Any ideas? >> >> regards >> Helmut >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > 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 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100616/a654688c/attachment.html From tony.tin at noahmedia.com.hk Wed Jun 16 19:26:54 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Thu, 17 Jun 2010 10:26:54 +0800 Subject: [Freeswitch-users] what's the cause of ozmod_isdn.c:1839 D-Chan Read Error! Message-ID: Hi, I've Digium TE220p and 4ess ISDN with openzap native pri stack. I found there are a lot of errors as below in the freeswitch.log. It seems that it doesn't affect the operation of system at the beginning, but finally freeswitch stop answering call. Could anyone help me. Thanks. 1. What's the cause of this error message. 2. How to avoid. 2010-06-16 10:07:04.945026 [ERR] ozmod_isdn.c:1839 D-Chan Read Error! Regards, Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/7d195a43/attachment.html From lakindia89 at gmail.com Wed Jun 16 21:03:07 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Thu, 17 Jun 2010 09:33:07 +0530 Subject: [Freeswitch-users] what's the cause of ozmod_isdn.c:1839 D-Chan Read Error! In-Reply-To: References: Message-ID: Please pastebin the entire freeswitch log, openzap.conf, openzap.conf.xml, so that someone can look into it and respond. On Thu, Jun 17, 2010 at 7:56 AM, Tony Tin wrote: > Hi, > > I've Digium TE220p and 4ess ISDN with openzap native pri stack. I found > there are a lot of errors as below in the freeswitch.log. It seems that it > doesn't affect the operation of system at the beginning, but finally > freeswitch stop answering call. Could anyone help me. Thanks. > > 1. What's the cause of this error message. > 2. How to avoid. > > 2010-06-16 10:07:04.945026 [ERR] ozmod_isdn.c:1839 D-Chan Read Error! > > > Regards, > Tony > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/9027506f/attachment.html From msc at freeswitch.org Wed Jun 16 21:45:07 2010 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 16 Jun 2010 21:45:07 -0700 Subject: [Freeswitch-users] what's the cause of ozmod_isdn.c:1839 D-Chan Read Error! In-Reply-To: References: Message-ID: <54B20F27-8B3F-4AD2-B45F-302A3759A25B@freeswitch.org> Native oz PRI stack does not explicitly support 4ess but does have 5ess. You might want to try libpri to see if that makes a difference since it does do 4e. -MC Sent from my iPhone On Jun 16, 2010, at 7:26 PM, Tony Tin wrote: > Hi, > > I've Digium TE220p and 4ess ISDN with openzap native pri stack. I > found there are a lot of errors as below in the freeswitch.log. It > seems that it doesn't affect the operation of system at the > beginning, but finally freeswitch stop answering call. Could anyone > help me. Thanks. > > 1. What's the cause of this error message. > 2. How to avoid. > > 2010-06-16 10:07:04.945026 [ERR] ozmod_isdn.c:1839 D-Chan Read Error! > > > Regards, > Tony > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 abu.4000 at gmail.com Wed Jun 16 21:57:41 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Thu, 17 Jun 2010 10:27:41 +0530 Subject: [Freeswitch-users] Issue in fifo while running it using outbound socket with async Message-ID: Dear all , -- BEST REGARDS N.ABUBACKER SOFTWARE ENGINEER BK SYSTEMS (P) LTD CHENNAI-23 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/25e4814d/attachment.html From abu.4000 at gmail.com Wed Jun 16 22:01:44 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Thu, 17 Jun 2010 10:31:44 +0530 Subject: [Freeswitch-users] Issue in fifo Message-ID: Dear all, I am facing problem in fifo while running that using outbound sockets in an async mode, it seems like connecting with the agents but no voice trasmission happens between them , please let me know if some body faced this issue already , and also give me a suggestion whether it is ok to run the fifo using outbound sockets. my main requirement is to play a file with a regular intervals for a waiting customers , Thanks in Advance ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/3ada7fa6/attachment.html From benxmy at gmail.com Wed Jun 16 17:24:08 2010 From: benxmy at gmail.com (benxmy) Date: Wed, 16 Jun 2010 17:24:08 -0700 (PDT) Subject: [Freeswitch-users] Merge two calls Message-ID: <1276734248530-5188763.post@n2.nabble.com> Hi, I'm quite new to freeswitch and voip, so this may be in some way a noob question but I've dug through a lot of the freeswitch docs and done quite a bit of searching and haven't figured it out yet. We're creating a relatively straightforward VoIP system to be used entirely internally (ex: users can only connect with other registered users within our system) and we'd like to be able to combine combine two calls into a single audio stream to the user without the two calls hearing each other. For example, if I'm talking to Mike on line 1 and I'm talking to Erin on line 2, is there a way for me to hear both Erin and Mike simultaneously but for them not to hear each other? Alternatively, is there a straightforward way to simply merge the calls into a conference-type experience where we all hear each other without the user explicitly setting up a conference call? Any and all input is greatly appreciated, as I'm up to my ears in freeswitch but have approximately zero experience with it! Ben -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Merge-two-calls-tp5188763p5188763.html Sent from the freeswitch-users mailing list archive at Nabble.com. From bandwidth.user at gmail.com Wed Jun 16 20:33:52 2010 From: bandwidth.user at gmail.com (roy) Date: Thu, 17 Jun 2010 11:33:52 +0800 Subject: [Freeswitch-users] mod_pocketsphinx Message-ID: <4C1997A0.1080909@gmail.com> Hi, Trying to get mod_pocketsphinx working from the current git tree, FreeSWITCH version: 1.0.head (git-2629a57 2010-06-16 22-42-15 +0200) however, I got this error during startup: 2010-06-17 11:18:08.032645 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_pocketsphinx.so **/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol: ngram_model_get_counts** Lib paths were already defined in /etc/ld.so.conf as: include /usr/local/freeswitch/lib /usr/local/freeswitch/mod # ls -al mod/mod_pocketsphinx.* -rwxr-xr-x 1 root staff 1467 Jun 17 10:49 mod/mod_pocketsphinx.la -rwxr-xr-x 1 root staff 905769 Jun 17 10:49 mod/mod_pocketsphinx.so Was wondering if someone was able to get this working. Manually downloading the current pocketsphinx and sphinxbase and renaming them into 0.4.99 and rebuilding didn't help either. Would appreciate if anyone can point me into the right direction. TIA. Roy From mike at jerris.com Wed Jun 16 22:30:22 2010 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Jun 2010 01:30:22 -0400 Subject: [Freeswitch-users] Issue in fifo In-Reply-To: References: Message-ID: <47C1371E-0F01-49B0-B061-CA530B0F6CAD@jerris.com> you don't need socket at all to play regular messages to the waiting callers. Using outbound sockets AND fifo doesn't really make any sense at all. Check out the wiki for more information how to correctly configure mod_fifo: http://wiki.freeswitch.org/wiki/Mod_fifo#Setting_MOH_and_announce_sounds Mike On Jun 17, 2010, at 1:01 AM, Abubacker siddiq wrote: > Dear all, > I am facing problem in fifo while running that using outbound sockets in an async mode, > it seems like connecting with the agents but no voice trasmission happens between them , > please let me know if some body faced this issue already , and also give me a suggestion > whether it is ok to run the fifo using outbound sockets. > > my main requirement is to play a file with a regular intervals for a waiting customers , > Thanks in Advance ! > From jalsot at gmail.com Wed Jun 16 23:43:15 2010 From: jalsot at gmail.com (Tamas) Date: Thu, 17 Jun 2010 08:43:15 +0200 Subject: [Freeswitch-users] How to continue call even if recording fails? In-Reply-To: <20100616212006.GD11308@hijacked.us> References: <1276669102712-5185161.post@n2.nabble.com> <4C1909BC.5040105@gmail.com> <9989BE22-A48B-460D-92A2-F0DCB44D2572@freeswitch.org> <20100616212006.GD11308@hijacked.us> Message-ID: <4C19C403.7050105@gmail.com> Hello, This feature has been added by http://jira.freeswitch.org/browse/FSCORE-591 where an environment variable version was provided too. Tamas 2010-06-16 23:20 keltez?ssel, Andrew Thompson ?rta: > On Wed, Jun 16, 2010 at 12:33:42PM -0500, Brian West wrote: > >> Just to clarify the reason the call fails is for consistency... Your app is broken if it doesn't make sure the directory is there.... What if this was a case where the phone call could save a client from a HUGE lawsuit and your application failed to actually do anything? See the reasoning? >> >> > Conversely, what if you'd merely like to make a 'best effort' attempt to > record the call, because the call is making you lots of money or AVOID > this fabled huge lawsuit. Sometimes drives fill up, disks get unmounted, > etc and no dialplan magic is going to save you. > > Personally I hate it when FS drops a call because recording failed, or I > tried to uuid_broadcast a nonexistant file, or whatever. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 helmut.kuper at ewetel.de Thu Jun 17 00:07:53 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 17 Jun 2010 09:07:53 +0200 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: References: <4C12D0F7.5000300@ewetel.de> <4C13CAEC.3020005@ewetel.de> Message-ID: <4C19C9C9.1080806@ewetel.de> Hi Samuel, unfortunately I haven't, yet. I found that the "new_ESLconnection function doesn't like a php resource as parameter like a socket, but rather a file descriptor like a numeric pointer. Unfortunately I have no idea how to convert a socket to such a file descriptor in php. Maybe there is a function in the socket object but I don't think so. So far as we didn't solve this problem here, I using my own php functions to read and write from/to incomming connections from freeswitch like this: function fs_send($csock, $cmd) { if (strlen($cmd) < 1) { acd_log(LOG_ERR, "Empty command\n"); return false; } $clen=strlen($cmd); if ($bytes_written = socket_write($csock, "$cmd\n\n", $clen) === false) { acd_log(LOG_ERR, "Can't send cmd! Wrote $bytes_written bytes\n"); socket_write($csock, "\n\n"); return false; } return true; } function fs_recv($csock, &$rbuf) { $i=0; $buf=""; do { acd_log(LOG_DEBUG, "Reading from socket '$csock'.\n"); if (($tbuf = socket_read($csock, 10000)) === false) { acd_log(LOG_ERR, "Can't read from client socket! (".socket_last_error($csock)."/".socket_strerror(socket_last_error($csock)).")\n"); return false; } acd_log(LOG_DEBUG, "Read ".strlen($tbuf)." bytes from socket '$csock'.\n"); $buf .= $tbuf; $i++; } while (substr($buf, -2) != "\n\n" and $i <10); foreach (explode("\n", $buf) as $t) { list($key, $val) = explode(": ", $t); $rbuf[strtolower($key)]=$val; } return true; } acd_log() is just my own logging functions. you can replace it with echo() or so. You can build on this functions more abstract functions like fs_connect(), fs_sendRecv(), ... The command format is as described on FS wiki (http://wiki.freeswitch.org/wiki/Event_socket_outbound) That works quite good, but using the phpmod for esl would be more easy.. regards Helmut On 17.06.2010 03:50, Samuel Macedo wrote: > Hi Helmut, > > I got the same error: > "No matching function for overloaded 'new_ESLconnection' in > /usr/share/php/ESL.php" > > Have you solved the problem? > > Thanks, > -- > Samuel Macedo > > On 12 June 2010 14:59, Helmut Kuper wrote: > >> Hi Michael, >> >> ESL is built and installed, phpmod is built and installed. >> >> inbound works, test scripts works >> >> It is just the problem of passing an existing socket to new ESLconnection. >> A socket resource is denied, an integer is accepted but is rejected with >> "bad file descriptor" >> >> >> >> Am 12.06.2010 07:06, schrieb Michael Collins: >> >> Make sure the ESL is properly built and that the PHP mod is built and >> installed. If you did a "make current" recently the you'll need to rebuild >> your ESL stuff. >> -MC >> >> On Fri, Jun 11, 2010 at 5:12 PM, Helmut Kuper wrote: >> >>> Hello, >>> >>> >>> I try to setup a php daemon which uses ESL. >>> >>> I run the sample php scripts successfully (inbound). Now I want to have >>> it outbound to my php daemon socket. >>> >>> The forked child process which has to interact with incoming tcp >>> connection from FS is started successfully. So in this state I have the >>> client socket which I have to pass now somehow to ESLconnection I guess. >>> >>> FS ruby wiki gives this as an example: >>> >>> @con = ESL::ESLconnection.new(client_socket.fileno) >>> >>> >>> In php I try this: >>> >>> $con = new ESLconnection($csock); >>> >>> As a result I got this error: >>> >>> PHP Fatal error: No matching function for overloaded >>> 'new_ESLconnection' in /usr/share/pear/ESL.php on line 117 >>> >>> >>> Any ideas? >>> >>> regards >>> Helmut From abu.4000 at gmail.com Thu Jun 17 00:13:23 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Thu, 17 Jun 2010 12:43:23 +0530 Subject: [Freeswitch-users] Issue in fifo In-Reply-To: <47C1371E-0F01-49B0-B061-CA530B0F6CAD@jerris.com> References: <47C1371E-0F01-49B0-B061-CA530B0F6CAD@jerris.com> Message-ID: Yeah , I agree with you mike the reason why I go for the outbound was I need to tell the position in the fifo to the customer , and update its status periodically , for example : consider already 3 members are waiting in the queue named A,B,C when the D customer calls I should announce him that his position is 4 , when the A has been answered and hangup then I should announce the D as his position is 3. like wise I should announce for the B and C too. the announcement should have to be done periodically . Is this possible using this fifo_music or chime_list ? On Thu, Jun 17, 2010 at 11:00 AM, Michael Jerris wrote: > you don't need socket at all to play regular messages to the waiting > callers. Using outbound sockets AND fifo doesn't really make any sense at > all. Check out the wiki for more information how to correctly configure > mod_fifo: > > http://wiki.freeswitch.org/wiki/Mod_fifo#Setting_MOH_and_announce_sounds > > Mike > > > On Jun 17, 2010, at 1:01 AM, Abubacker siddiq wrote: > > > Dear all, > > I am facing problem in fifo while running that using outbound > sockets in an async mode, > > it seems like connecting with the agents but no voice trasmission happens > between them , > > please let me know if some body faced this issue already , and also give > me a suggestion > > whether it is ok to run the fifo using outbound sockets. > > > > my main requirement is to play a file with a regular intervals for a > waiting customers , > > Thanks in Advance ! > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 N.ABUBACKER SOFTWARE ENGINEER BK SYSTEMS (P) LTD CHENNAI-23 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/de988c15/attachment.html From helmut.kuper at ewetel.de Thu Jun 17 00:55:41 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 17 Jun 2010 09:55:41 +0200 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: <4C19C9C9.1080806@ewetel.de> References: <4C12D0F7.5000300@ewetel.de> <4C13CAEC.3020005@ewetel.de> <4C19C9C9.1080806@ewetel.de> Message-ID: <4C19D4FD.1050800@ewetel.de> Hi, a bit success now. When I use this command: $con = new ESLconnection(intval($csock)); I got an ESLconnection object, but it is not usable. var_dump($con) shows object(ESLconnection)#1 (1) { ["_cPtr"]=> resource(7) of type (_p_ESLconnection) } which looks good to me, but strace still shows "Bad file descriptor": setsockopt(6, SOL_TCP, TCP_NODELAY, [1], 4) = -1 EBADF (Bad file descriptor) sendto(6, "connect\n\n", 9, 0, NULL, 0) = -1 EBADF (Bad file descriptor) Calling $con->getINFO() results in NULL ... regards Helmut From samu60 at gmail.com Thu Jun 17 01:14:59 2010 From: samu60 at gmail.com (samuel) Date: Thu, 17 Jun 2010 10:14:59 +0200 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: References: <4C17B5BC.5050608@coppice.org> <4C18BEB7.5070702@coppice.org> Message-ID: Just created the bug MODENDP-312 with all the information. Thank you very much in advance, Samuel On 16 June 2010 18:30, Anthony Minessale wrote: > The entire log of the call with the debug all the way up including sip > traces would be a good start > > sofia profile internal siptrace on > console loglevel debug > > edit fax.conf.xml and set verbose to true and restart > > get a pcap of the whole thing. > > open an issue on jira and attach everything and report the bug ID here. > > > > On Wed, Jun 16, 2010 at 9:15 AM, samuel wrote: > >> i've provided all the information i've got and what the other point has >> reported. >> >> What extra information do you need? >> >> thanks a lot, >> Samuel. >> >> >> On 16 June 2010 14:08, Steve Underwood wrote: >> >>> On 06/16/2010 07:44 PM, samuel wrote: >>> > I'm trying yo know what the remote end point brand/model/firmware/??? >>> > is but I could not get this information yet. >>> > The other point says that after the T.38 SDP negotiation, there must >>> > be a "negotiation" (after the ACK of the T38 renegotiation there must >>> > be an exchange of Send preamble; DIS (and later DCS) that never >>> > happens...) and freeswitch does not start it and that is why the >>> > transmission results in a call drop after 25 seconds. >>> > >>> > >>> > Thank you very much for your answers so far, >>> > Samuel >>> >>> So far you have kept that much secret. What you supplied was only the >>> very last stages of a call log, where Freeswitch seems to be output the >>> messages it should, but gets nothing in return. >>> >>> Steve >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > > -- > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/810c356c/attachment.html From mike at jerris.com Thu Jun 17 01:26:05 2010 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Jun 2010 04:26:05 -0400 Subject: [Freeswitch-users] Issue in fifo In-Reply-To: References: <47C1371E-0F01-49B0-B061-CA530B0F6CAD@jerris.com> Message-ID: <8BA757CF-CEBE-4EF1-A16F-B2C91E6BF199@jerris.com> Outbound still makes no sense at all here. Can't do 2 things at once in one thread. Your call is either running the socket app for outbound, or running the queue. inbound socket would probably make sense here if you need to do something like that. Also, when a priority customer calls in, do you tell the guy that was previously #3 in queue the next time around that he is #4? Mike On Jun 17, 2010, at 3:13 AM, Abubacker siddiq wrote: > Yeah , I agree with you mike the reason why I go for the outbound was I need to tell the position > in the fifo to the customer , and update its status periodically , > for example : > consider already 3 members are waiting in the queue named A,B,C > when the D customer calls I should announce him that his position is 4 , > when the A has been answered and hangup then I should announce the D as his position > is 3. like wise I should announce for the B and C too. > > the announcement should have to be done periodically . > > Is this possible using this fifo_music or chime_list ? > > > On Thu, Jun 17, 2010 at 11:00 AM, Michael Jerris wrote: > you don't need socket at all to play regular messages to the waiting callers. Using outbound sockets AND fifo doesn't really make any sense at all. Check out the wiki for more information how to correctly configure mod_fifo: > > http://wiki.freeswitch.org/wiki/Mod_fifo#Setting_MOH_and_announce_sounds > > Mike > > > On Jun 17, 2010, at 1:01 AM, Abubacker siddiq wrote: > > > Dear all, > > I am facing problem in fifo while running that using outbound sockets in an async mode, > > it seems like connecting with the agents but no voice trasmission happens between them , > > please let me know if some body faced this issue already , and also give me a suggestion > > whether it is ok to run the fifo using outbound sockets. > > > > my main requirement is to play a file with a regular intervals for a waiting customers , > > Thanks in Advance ! > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > N.ABUBACKER > SOFTWARE ENGINEER > BK SYSTEMS (P) LTD > CHENNAI-23 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/87252a56/attachment.html From abu.4000 at gmail.com Thu Jun 17 01:55:18 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Thu, 17 Jun 2010 14:25:18 +0530 Subject: [Freeswitch-users] Issue in fifo In-Reply-To: <8BA757CF-CEBE-4EF1-A16F-B2C91E6BF199@jerris.com> References: <47C1371E-0F01-49B0-B061-CA530B0F6CAD@jerris.com> <8BA757CF-CEBE-4EF1-A16F-B2C91E6BF199@jerris.com> Message-ID: Thanks Mike , I have done using inbound sockets. No currently I have no idea regarding priority customer and all that. Can you tell me what exactly happens when I use fifo using outbound async sockets, thanks once again. On 6/17/10, Michael Jerris wrote: > Outbound still makes no sense at all here. Can't do 2 things at once in one > thread. Your call is either running the socket app for outbound, or running > the queue. inbound socket would probably make sense here if you need to do > something like that. Also, when a priority customer calls in, do you tell > the guy that was previously #3 in queue the next time around that he is #4? > > Mike > > On Jun 17, 2010, at 3:13 AM, Abubacker siddiq wrote: > >> Yeah , I agree with you mike the reason why I go for the outbound was I >> need to tell the position >> in the fifo to the customer , and update its status periodically , >> for example : >> consider already 3 members are waiting in the queue named A,B,C >> when the D customer calls I should announce him that his position is 4 , >> when the A has been answered and hangup then I should announce the D as >> his position >> is 3. like wise I should announce for the B and C too. >> >> the announcement should have to be done periodically . >> >> Is this possible using this fifo_music or chime_list ? >> >> >> On Thu, Jun 17, 2010 at 11:00 AM, Michael Jerris wrote: >> you don't need socket at all to play regular messages to the waiting >> callers. Using outbound sockets AND fifo doesn't really make any sense at >> all. Check out the wiki for more information how to correctly configure >> mod_fifo: >> >> http://wiki.freeswitch.org/wiki/Mod_fifo#Setting_MOH_and_announce_sounds >> >> Mike >> >> >> On Jun 17, 2010, at 1:01 AM, Abubacker siddiq wrote: >> >> > Dear all, >> > I am facing problem in fifo while running that using outbound >> > sockets in an async mode, >> > it seems like connecting with the agents but no voice trasmission >> > happens between them , >> > please let me know if some body faced this issue already , and also give >> > me a suggestion >> > whether it is ok to run the fifo using outbound sockets. >> > >> > my main requirement is to play a file with a regular intervals for a >> > waiting customers , >> > Thanks in Advance ! >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 >> N.ABUBACKER >> SOFTWARE ENGINEER >> BK SYSTEMS (P) LTD >> CHENNAI-23 >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 N.ABUBACKER SOFTWARE ENGINEER BK SYSTEMS (P) LTD CHENNAI-23 From nagalenoj at gmail.com Thu Jun 17 02:23:39 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Thu, 17 Jun 2010 14:53:39 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: Anthony, But, then there is no use. Am I right? Usually, we'll use the group_confirm_cancel_timeout only when we need to override the leg_timeout. But it happens in reverse in this case., I've tried using the group_confirm_cancel_timeout along with call_timeout and things happening similar like setting leg_timout. Then, tried without setting leg_timeout and call_timeout explicitly. * In this case if the callee doesn't picks the call, it disconnects the leg in 30 secs. * If he answers the call and the script continues to execute, the leg is disconnected in 60 secs. What I need to do is, when the callee picks the call the leg_timeout should not be accounted more and the leg shouldn't be disconnected because of leg_timeout after that. Any other way of doing this?! On Tue, Jun 15, 2010 at 10:53 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > leg timeout beats the group confirm timeouts > > > On Tue, Jun 15, 2010 at 12:28 AM, Nagalenoj H. wrote: > >> Dear friends, >> I've tried using the group_confirm_cancel_timeout channel variable. >> I've written a testing script to get digits before bridging. But, it doesn't >> seem to be working. >> >> My understanding after reading wiki is, >> * When I dial [leg_timeout=10]user/1005, if he answers before >> timeout and in the process of giving digits, then the call shouldn't be >> disconnected after the leg_timeout secs (10 sec in the example). >> >> But, When I experiment it, the call is getting disconnected after 10 >> seconds and it doesn't bother whether the callee has answered the >> call(Started giving digits) or not answered at all. >> >> I've checked it with nc as follows, >> >> sendmsg >> call-command: execute >> execute-app-name: set >> execute-app-arg: group_confirm_key=exec >> >> sendmsg >> call-command: execute >> execute-app-name: set >> execute-app-arg: group_confirm_file=perl /root/confirm.pl >> >> sendmsg >> call-command: execute >> execute-app-name: set >> execute-app-arg: group_confirm_cancel_timeout=1 >> >> sendmsg >> call-command: execute >> execute-app-name: bridge >> execute-app-arg: [leg_timeout=10]user/1005 >> >> And here is the script, >> >> use freeswitch; >> our $session; >> my $digit; >> >> while(1) { >> # Wait till response timeout for the first digit. >> $digit = $session->getDigits(1, "", 10000); >> freeswitch::consoleLog ("info","Digit>>".$digit."<<"); >> >> if (! $session->ready() ) { >> freeswitch::consoleLog("info","Going to Exit\n"); >> last; >> } >> if (defined $digit and $digit ne "" ) { >> freeswitch::consoleLog("info","DTMF received: $digit\n"); >> if ($digit eq '#') { >> return; >> } >> } >> else { >> freeswitch::consoleLog("info","Timeout\n"); >> $session->hangup(); >> } >> } >> 1; >> >> If my understanding is right then, I believe there is something wrong with >> channel_variable. >> >> Kindly help me to resolve this. >> >> -- >> Regards, >> Nagalenoj H. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/db64cf86/attachment-0001.html From craig at overthewire.com.au Thu Jun 17 04:15:53 2010 From: craig at overthewire.com.au (Craig Askings) Date: Thu, 17 Jun 2010 21:15:53 +1000 Subject: [Freeswitch-users] [Freeswitch-dev] Special Announcement: Be On The Community Conf Call Tomorrow! In-Reply-To: References: <663F299A58AC4E9099617A59AB3E5BEC@dell9400> Message-ID: A copy of the beer source and one example implementation given to every early bird registrant? Craig. On 16 June 2010 10:13, Brian West wrote: > Nope.... Keep guessing... its ClueCon Related.... so get your credit cards > ready! ;) > > /b > > On Jun 15, 2010, at 7:06 PM, Jan Berger wrote: > > > Oh ? is FS running on hes iPad? > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/a0d24b1d/attachment.html From stephen at stephenjc.com Thu Jun 17 04:34:44 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Thu, 17 Jun 2010 07:34:44 -0400 Subject: [Freeswitch-users] G729 Licenses Message-ID: I had some questions about the G729 licenses. 1- Can they be shared between installs? 2- Can licenses be moved? 2- If both my A and B legs are G729 will a license only be used when they are using IVR? 3- In the same scenario above will a license be used if i am recording a call. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/2b3ec9e6/attachment.html From rupa at rupa.com Thu Jun 17 05:24:10 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 17 Jun 2010 07:24:10 -0500 Subject: [Freeswitch-users] G729 Licenses In-Reply-To: References: Message-ID: 1) no 2) yes, but it is not just "move them" -- you have to fill out a form documenting that old equipment is decomissioned and are limited to the max times you move 3) a licence will be used for each encode/decode channel needed. a/b being g729 with no other requirements will not use a licence. Anything that uses a media bug (eg: recording, eavesdrop, tone detect) or that interacts directly with the media (voicemail, ivr) will eat licenses. If you are eavesdropping with a g729 endpoint that'll eat another. On Thu, Jun 17, 2010 at 6:34 AM, stephen at stephenjc wrote: > I had some questions about the G729 licenses. > > 1- Can they be shared between installs? > 2- Can licenses be moved? > 2- If both my A and B legs are G729 will a license only be used when they > are using IVR? > 3- In the same scenario above will a license be used if i am recording a > call. > > > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/ee4d544b/attachment.html From brian at freeswitch.org Thu Jun 17 05:34:56 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Jun 2010 07:34:56 -0500 Subject: [Freeswitch-users] mod_pocketsphinx In-Reply-To: <4C1997A0.1080909@gmail.com> References: <4C1997A0.1080909@gmail.com> Message-ID: your running ubuntu aren't you? /b On Jun 16, 2010, at 10:33 PM, roy wrote: > 2010-06-17 11:18:08.032645 [CRIT] switch_loadable_module.c:882 Error > Loading module /usr/local/freeswitch/mod/mod_pocketsphinx.so > **/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol: > ngram_model_get_counts** From stephen at stephenjc.com Thu Jun 17 05:46:53 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Thu, 17 Jun 2010 08:46:53 -0400 Subject: [Freeswitch-users] G729 Licenses In-Reply-To: References: Message-ID: for #3, if both legs are g729 will the license be returned when the media bug is stopped or when the call is hungup? Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print On Thu, Jun 17, 2010 at 8:24 AM, Rupa Schomaker wrote: > 1) no > 2) yes, but it is not just "move them" -- you have to fill out a form > documenting that old equipment is decomissioned and are limited to the max > times you move > 3) a licence will be used for each encode/decode channel needed. a/b being > g729 with no other requirements will not use a licence. Anything that uses > a media bug (eg: recording, eavesdrop, tone detect) or that interacts > directly with the media (voicemail, ivr) will eat licenses. If you are > eavesdropping with a g729 endpoint that'll eat another. > > On Thu, Jun 17, 2010 at 6:34 AM, stephen at stephenjc < > stephen at stephenjc.com> wrote: > >> I had some questions about the G729 licenses. >> >> 1- Can they be shared between installs? >> 2- Can licenses be moved? >> 2- If both my A and B legs are G729 will a license only be used when they >> are using IVR? >> 3- In the same scenario above will a license be used if i am recording a >> call. >> >> >> >> Thanks, >> Stephen C >> -All of my email addresses go to the same place >> -Save Paper, think before you print >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/7ddb3c91/attachment.html From brian at freeswitch.org Thu Jun 17 05:50:07 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Jun 2010 07:50:07 -0500 Subject: [Freeswitch-users] G729 Licenses In-Reply-To: References: Message-ID: Yes. /b On Jun 17, 2010, at 7:46 AM, stephen at stephenjc wrote: > for #3, if both legs are g729 will the license be returned when the media bug is stopped or when the call is hungup? > > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print From xengelpublicx at gmail.com Thu Jun 17 05:57:30 2010 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Thu, 17 Jun 2010 16:57:30 +0400 Subject: [Freeswitch-users] presence in linksys spa932 Message-ID: I found a bug in the presence of the linksys spa932 (configure this article: http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932). If during the conversation (lamp on) to the subscriber, someone called, the lamp on the panel goes into the state off. Who can verify whether he has reproduced the same problem? Unit key: fnc=blf+sd+cp;sub=110@$PROXY freeswitch 1.0.6 (git 10 06 2010) Linksys spa962: Software Version: 6.1.3(a) Hardware Version: 1.0.3(917f) Linksys spa932 Unit Enable: Yes Unit Online: Yes Subscribe Expires: 600 Subscribe Retry Interval: 6 HW Version: 1.0.6 SW Version: 2.0.2 Configure image: http://img192.imageshack.us/img192/2320/linksysspa932.png -- Best regards, Vladimir Elizarov From dswardstrom at remotelink.com Thu Jun 17 07:41:06 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Thu, 17 Jun 2010 09:41:06 -0500 (CDT) Subject: [Freeswitch-users] Javascript session.sayphrase() In-Reply-To: <1903777814.103.1276785654549.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <112777075.105.1276785666670.JavaMail.root@srvr12.remotelinkml.com> I have a need to provide date & time to a caller. The current application allows for interruption via a DTMF key. It would be nice to use session.sayphrase() but the Wiki seems to indicate that this does not allow operation in an interruptable way. Is this correct or do I need to go to lower levels and perhaps change xml files for the "Phrases Section"? so that a play can (optionally) use the dtmf detection and callback arguments? Regards, Paul David Swardstrom From anthony.minessale at gmail.com Thu Jun 17 07:52:21 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Jun 2010 09:52:21 -0500 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: no there is no way, besides making both timeouts longer. you could file a feature request/bounty to ask for a feature to stop the leg timer when you reach the confirm. On Thu, Jun 17, 2010 at 4:23 AM, Nagalenoj H. wrote: > Anthony, > But, then there is no use. Am I right? Usually, we'll use the > group_confirm_cancel_timeout only when we need to override the leg_timeout. > But it happens in reverse in this case., > > I've tried using the group_confirm_cancel_timeout along with call_timeout > and things happening similar like setting leg_timout. > > Then, tried without setting leg_timeout and call_timeout explicitly. > * In this case if the callee doesn't picks the call, it disconnects > the leg in 30 secs. > * If he answers the call and the script continues to execute, the > leg is disconnected in 60 secs. > > What I need to do is, when the callee picks the call the leg_timeout should > not be accounted more and the leg shouldn't be disconnected because of > leg_timeout after that. > > Any other way of doing this?! > > > > On Tue, Jun 15, 2010 at 10:53 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> leg timeout beats the group confirm timeouts >> >> >> On Tue, Jun 15, 2010 at 12:28 AM, Nagalenoj H. wrote: >> >>> Dear friends, >>> I've tried using the group_confirm_cancel_timeout channel variable. >>> I've written a testing script to get digits before bridging. But, it doesn't >>> seem to be working. >>> >>> My understanding after reading wiki is, >>> * When I dial [leg_timeout=10]user/1005, if he answers before >>> timeout and in the process of giving digits, then the call shouldn't be >>> disconnected after the leg_timeout secs (10 sec in the example). >>> >>> But, When I experiment it, the call is getting disconnected after 10 >>> seconds and it doesn't bother whether the callee has answered the >>> call(Started giving digits) or not answered at all. >>> >>> I've checked it with nc as follows, >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: set >>> execute-app-arg: group_confirm_key=exec >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: set >>> execute-app-arg: group_confirm_file=perl /root/confirm.pl >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: set >>> execute-app-arg: group_confirm_cancel_timeout=1 >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: bridge >>> execute-app-arg: [leg_timeout=10]user/1005 >>> >>> And here is the script, >>> >>> use freeswitch; >>> our $session; >>> my $digit; >>> >>> while(1) { >>> # Wait till response timeout for the first digit. >>> $digit = $session->getDigits(1, "", 10000); >>> freeswitch::consoleLog ("info","Digit>>".$digit."<<"); >>> >>> if (! $session->ready() ) { >>> freeswitch::consoleLog("info","Going to Exit\n"); >>> last; >>> } >>> if (defined $digit and $digit ne "" ) { >>> freeswitch::consoleLog("info","DTMF received: $digit\n"); >>> if ($digit eq '#') { >>> return; >>> } >>> } >>> else { >>> freeswitch::consoleLog("info","Timeout\n"); >>> $session->hangup(); >>> } >>> } >>> 1; >>> >>> If my understanding is right then, I believe there is something wrong >>> with channel_variable. >>> >>> Kindly help me to resolve this. >>> >>> -- >>> Regards, >>> Nagalenoj H. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100617/e991efd3/attachment.html From errotan at elder.hu Thu Jun 17 07:56:06 2010 From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=) Date: Thu, 17 Jun 2010 16:56:06 +0200 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <1276734248530-5188763.post@n2.nabble.com> References: <1276734248530-5188763.post@n2.nabble.com> Message-ID: <201006171656.06384.errotan@elder.hu> 2010. j?nius 17. 02.24.08 d?tummal benxmy az al?bbiakat ?rta: > Hi, > I'm quite new to freeswitch and voip, so this may be in some way a noob > question but I've dug through a lot of the freeswitch docs and done quite a > bit of searching and haven't figured it out yet. > > We're creating a relatively straightforward VoIP system to be used entirely > internally (ex: users can only connect with other registered users within > our system) and we'd like to be able to combine combine two calls into a > single audio stream to the user without the two calls hearing each other. > For example, if I'm talking to Mike on line 1 and I'm talking to Erin on > line 2, is there a way for me to hear both Erin and Mike simultaneously but > for them not to hear each other? > > Alternatively, is there a straightforward way to simply merge the calls > into a conference-type experience where we all hear each other without the > user explicitly setting up a conference call? > > Any and all input is greatly appreciated, as I'm up to my ears in > freeswitch but have approximately zero experience with it! > > Ben Hi. What is the point of talking to 2 person while they can't hear each other ? For example when you say a sentence to person "A" and he replies back with lots of sentences how person "B" knows when he can talk if he can't hear person "A"? Person "B" starts talking while person "A" so you can't understand a word. This would only confuse people... From samu60 at gmail.com Thu Jun 17 09:16:18 2010 From: samu60 at gmail.com (samuel) Date: Thu, 17 Jun 2010 18:16:18 +0200 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: References: <4C17B5BC.5050608@coppice.org> <4C18BEB7.5070702@coppice.org> Message-ID: Just to let you know that with latest changes in head version done by Michael, everything seems to be working and I'm able to receive faxes via T.38. Thank you very much and specially to Michael that took reacted to the bug in less than 24hours and solved it....great project and amazing comunity!!! Samuel On 17 June 2010 10:14, samuel wrote: > Just created the bug MODENDP-312 with all the information. > > Thank you very much in advance, > Samuel > > > > On 16 June 2010 18:30, Anthony Minessale wrote: > >> The entire log of the call with the debug all the way up including sip >> traces would be a good start >> >> sofia profile internal siptrace on >> console loglevel debug >> >> edit fax.conf.xml and set verbose to true and restart >> >> get a pcap of the whole thing. >> >> open an issue on jira and attach everything and report the bug ID here. >> >> >> >> On Wed, Jun 16, 2010 at 9:15 AM, samuel wrote: >> >>> i've provided all the information i've got and what the other point has >>> reported. >>> >>> What extra information do you need? >>> >>> thanks a lot, >>> Samuel. >>> >>> >>> On 16 June 2010 14:08, Steve Underwood wrote: >>> >>>> On 06/16/2010 07:44 PM, samuel wrote: >>>> > I'm trying yo know what the remote end point brand/model/firmware/??? >>>> > is but I could not get this information yet. >>>> > The other point says that after the T.38 SDP negotiation, there must >>>> > be a "negotiation" (after the ACK of the T38 renegotiation there must >>>> > be an exchange of Send preamble; DIS (and later DCS) that never >>>> > happens...) and freeswitch does not start it and that is why the >>>> > transmission results in a call drop after 25 seconds. >>>> > >>>> > >>>> > Thank you very much for your answers so far, >>>> > Samuel >>>> >>>> So far you have kept that much secret. What you supplied was only the >>>> very last stages of a call log, where Freeswitch seems to be output the >>>> messages it should, but gets nothing in return. >>>> >>>> Steve >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> >> >> -- >> 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/563ed25e/attachment-0001.html From samu60 at gmail.com Thu Jun 17 09:18:43 2010 From: samu60 at gmail.com (samuel) Date: Thu, 17 Jun 2010 18:18:43 +0200 Subject: [Freeswitch-users] T38 unsuccessful: result (48) Disconnected after permitted retries In-Reply-To: References: <4C17B5BC.5050608@coppice.org> <4C18BEB7.5070702@coppice.org> Message-ID: sorry r/Michael/Antony On 17 June 2010 18:16, samuel wrote: > Just to let you know that with latest changes in head version done by > Michael, everything seems to be working and I'm able to receive faxes via > T.38. > > Thank you very much and specially to Michael that took reacted to the bug > in less than 24hours and solved it....great project and amazing comunity!!! > > Samuel > > > On 17 June 2010 10:14, samuel wrote: > >> Just created the bug MODENDP-312 with all the information. >> >> Thank you very much in advance, >> Samuel >> >> >> >> On 16 June 2010 18:30, Anthony Minessale wrote: >> >>> The entire log of the call with the debug all the way up including sip >>> traces would be a good start >>> >>> sofia profile internal siptrace on >>> console loglevel debug >>> >>> edit fax.conf.xml and set verbose to true and restart >>> >>> get a pcap of the whole thing. >>> >>> open an issue on jira and attach everything and report the bug ID here. >>> >>> >>> >>> On Wed, Jun 16, 2010 at 9:15 AM, samuel wrote: >>> >>>> i've provided all the information i've got and what the other point has >>>> reported. >>>> >>>> What extra information do you need? >>>> >>>> thanks a lot, >>>> Samuel. >>>> >>>> >>>> On 16 June 2010 14:08, Steve Underwood wrote: >>>> >>>>> On 06/16/2010 07:44 PM, samuel wrote: >>>>> > I'm trying yo know what the remote end point brand/model/firmware/??? >>>>> > is but I could not get this information yet. >>>>> > The other point says that after the T.38 SDP negotiation, there must >>>>> > be a "negotiation" (after the ACK of the T38 renegotiation there must >>>>> > be an exchange of Send preamble; DIS (and later DCS) that never >>>>> > happens...) and freeswitch does not start it and that is why the >>>>> > transmission results in a call drop after 25 seconds. >>>>> > >>>>> > >>>>> > Thank you very much for your answers so far, >>>>> > Samuel >>>>> >>>>> So far you have kept that much secret. What you supplied was only the >>>>> very last stages of a call log, where Freeswitch seems to be output the >>>>> messages it should, but gets nothing in return. >>>>> >>>>> Steve >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.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 >>>> >>>> >>> >>> >>> -- >>> 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/cb1c430b/attachment.html From anthony.minessale at gmail.com Thu Jun 17 10:05:37 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Jun 2010 12:05:37 -0500 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <201006171656.06384.errotan@elder.hu> References: <1276734248530-5188763.post@n2.nabble.com> <201006171656.06384.errotan@elder.hu> Message-ID: I am going to guess 911 callcenter where Mike is bleeding somewhere and Erin is a police dispatcher. On Thu, Jun 17, 2010 at 9:56 AM, Pusk?s Zsolt wrote: > 2010. j?nius 17. 02.24.08 d?tummal benxmy az al?bbiakat ?rta: > > Hi, > > I'm quite new to freeswitch and voip, so this may be in some way a noob > > question but I've dug through a lot of the freeswitch docs and done quite > a > > bit of searching and haven't figured it out yet. > > > > We're creating a relatively straightforward VoIP system to be used > entirely > > internally (ex: users can only connect with other registered users within > > our system) and we'd like to be able to combine combine two calls into a > > single audio stream to the user without the two calls hearing each other. > > For example, if I'm talking to Mike on line 1 and I'm talking to Erin on > > line 2, is there a way for me to hear both Erin and Mike simultaneously > but > > for them not to hear each other? > > > > Alternatively, is there a straightforward way to simply merge the calls > > into a conference-type experience where we all hear each other without > the > > user explicitly setting up a conference call? > > > > Any and all input is greatly appreciated, as I'm up to my ears in > > freeswitch but have approximately zero experience with it! > > > > Ben > > Hi. > > What is the point of talking to 2 person while they can't hear each other ? > For example when you say a sentence to person "A" and he replies back with > lots of sentences how person "B" knows when he can talk if he can't hear > person "A"? Person "B" starts talking while person "A" so you can't > understand > a word. This would only confuse people... > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100617/103787e3/attachment.html From msc at freeswitch.org Thu Jun 17 10:22:46 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Jun 2010 10:22:46 -0700 Subject: [Freeswitch-users] Javascript session.sayphrase() In-Reply-To: <112777075.105.1276785666670.JavaMail.root@srvr12.remotelinkml.com> References: <1903777814.103.1276785654549.JavaMail.root@srvr12.remotelinkml.com> <112777075.105.1276785666670.JavaMail.root@srvr12.remotelinkml.com> Message-ID: Just use playAndGetDigits. -MC On Thu, Jun 17, 2010 at 7:41 AM, David Swardstrom < dswardstrom at remotelink.com> wrote: > I have a need to provide date & time to a caller. > The current application allows for interruption via a DTMF key. > > It would be nice to use session.sayphrase() but the Wiki seems to indicate > that this does not allow operation in an interruptable way. > > Is this correct or do I need to go to lower levels and perhaps change > xml files for the "Phrases Section"? so that a play can (optionally) > use the dtmf detection and callback arguments? > > Regards, > Paul David Swardstrom > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/754b2913/attachment.html From sos at sokhapkin.dyndns.org Thu Jun 17 10:24:09 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Thu, 17 Jun 2010 13:24:09 -0400 Subject: [Freeswitch-users] Merge two calls In-Reply-To: References: <1276734248530-5188763.post@n2.nabble.com> <201006171656.06384.errotan@elder.hu> Message-ID: <201006171324.09460.sos@sokhapkin.dyndns.org> It's a good idea. 911 at freeswitch.org SIP URI with $1 per minute rate :-D I want to thank you again for providing a really good software. Sometime very buggy, but overall it's very good. I will never return to asterisk. On Thursday 17 June 2010, Anthony Minessale wrote: > I am going to guess 911 callcenter where Mike is bleeding somewhere and > Erin is a police dispatcher. > > On Thu, Jun 17, 2010 at 9:56 AM, Pusk?s Zsolt wrote: > > 2010. j?nius 17. 02.24.08 d?tummal benxmy az al?bbiakat ?rta: > > > Hi, > > > I'm quite new to freeswitch and voip, so this may be in some way a noob > > > question but I've dug through a lot of the freeswitch docs and done > > > quite > > > > a > > > > > bit of searching and haven't figured it out yet. > > > > > > We're creating a relatively straightforward VoIP system to be used > > > > entirely > > > > > internally (ex: users can only connect with other registered users > > > within our system) and we'd like to be able to combine combine two > > > calls into a single audio stream to the user without the two calls > > > hearing each other. For example, if I'm talking to Mike on line 1 and > > > I'm talking to Erin on line 2, is there a way for me to hear both Erin > > > and Mike simultaneously > > > > but > > > > > for them not to hear each other? > > > > > > Alternatively, is there a straightforward way to simply merge the calls > > > into a conference-type experience where we all hear each other without > > > > the > > > > > user explicitly setting up a conference call? > > > > > > Any and all input is greatly appreciated, as I'm up to my ears in > > > freeswitch but have approximately zero experience with it! > > > > > > Ben > > > > Hi. > > > > What is the point of talking to 2 person while they can't hear each other > > ? For example when you say a sentence to person "A" and he replies back > > with lots of sentences how person "B" knows when he can talk if he can't > > hear person "A"? Person "B" starts talking while person "A" so you can't > > understand > > a word. This would only confuse people... > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 kouxiaodong at gmail.com Thu Jun 17 01:40:47 2010 From: kouxiaodong at gmail.com (k xd) Date: Thu, 17 Jun 2010 16:40:47 +0800 Subject: [Freeswitch-users] How to enable mod_skypopen without sound card Message-ID: Hi, I setup freeswitch in a window server and then startup 2 skype clients, however when I call registered skype user, I got this message "Contact can only receive IMs". The server doesn't have sound card. So I guess that when registered skype users try to transfer the incoming call, because of no sound card it terminated the call directly. Does anyone know how to solve this issue? Thanks, Will -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/ac81f50b/attachment.html From wespear at gmail.com Thu Jun 17 10:32:37 2010 From: wespear at gmail.com (Wes Pearce) Date: Thu, 17 Jun 2010 10:32:37 -0700 (PDT) Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? Message-ID: <1276795957115-5191927.post@n2.nabble.com> Hello Users, I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in the office here. I've been trying to get SCA working unsuccessfully. While I was digging around, I found this thread: http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html Which describes my exact problem. SCA works neatly on outbound calls, and SCA works on incoming calls until the phone is answered... at which point the call dissapears from the other lines. Brian West mentioned this has something to do with Aastras inconsistent call-info headers. I've been speaking with a support representative from Aastra named Brian Epps, who says their engineers are unaware of any SCA problems. Here are the appropriate SIP traces: incoming call: http://pastebin.com/QK9K3KL9 outgoing call: http://pastebin.com/zzxbm9mq There are call info headers in both traces. It's just the call-info headers in the incoming trace look incorrect. I'm wondering if there has been any movement on this, or if I've just screwed something up somewhere. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp5191927p5191927.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gmaruzz at celliax.org Thu Jun 17 10:52:22 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 17 Jun 2010 19:52:22 +0200 Subject: [Freeswitch-users] How to enable mod_skypopen without sound card In-Reply-To: References: Message-ID: On Thu, Jun 17, 2010 at 10:40 AM, k xd wrote: > Hi, > I setup freeswitch in a window server and then startup 2 skype clients, > however when I call registered skype user, I got this message "Contact can > only receive IMs". The server doesn't have sound card. So I guess that when > registered skype users try to transfer the incoming call, because of no > sound card it terminated the call directly. > Does anyone know how to solve this issue? A windows server has no way (that I know) to fake a soundcard (as opposed to the snd-dummy "fake" audio driver in Linux), and the Skype clients check if there is a soundcard available, and refuse to do audio if a soundcard is not available (is stupid, but is like that). So, your only option is to add a soundcard to the server, a cheap dongle format USB soundcard would do as well (the soundcard is not used for anything, but the Skype client check about its existance). -giovanni > Thanks, > Will > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From testeador01 at gmail.com Thu Jun 17 10:56:46 2010 From: testeador01 at gmail.com (Milena) Date: Thu, 17 Jun 2010 12:56:46 -0500 Subject: [Freeswitch-users] Issue with an IVR-menu Message-ID: Hello, Does anybody know why would freeswitch say ... switch_ivr_menu.c:851 Unable to build xml menu ... mod_dptools.c:1264 Unable to create menu when trying to call an IVR that was working fine before I updated this morning? ... and how to fix it? ;) http://pastebin.freeswitch.org/13209 thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/9eaec6b2/attachment.html From brian at freeswitch.org Thu Jun 17 11:12:43 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Jun 2010 13:12:43 -0500 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: <1276795957115-5191927.post@n2.nabble.com> References: <1276795957115-5191927.post@n2.nabble.com> Message-ID: <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> The issue is the absence of the extra bits in the call-info header on every packet involved in the call transaction... but its worse than that if you want SCA you can only have ONE account on the phone and HOPE and pray it behaves. I have tried to get Aastra to fix it but as far as I'm concerned your best bet is to call Polycom and trade all those Aastra's for Polycom's they have a trade up program and all you have to do is take a baseball bat and make the Aastra's inoperable or not they seems to have the inoperable part covered already right out of the box. /b On Jun 17, 2010, at 12:32 PM, Wes Pearce wrote: > > Hello Users, > > I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in the > office here. I've been trying to get SCA working unsuccessfully. While I was > digging around, I found this thread: > > http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html > > Which describes my exact problem. SCA works neatly on outbound calls, and > SCA works on incoming calls until the phone is answered... at which point > the call dissapears from the other lines. > > Brian West mentioned this has something to do with Aastras inconsistent > call-info headers. > > I've been speaking with a support representative from Aastra named Brian > Epps, who says their engineers are unaware of any SCA problems. > > Here are the appropriate SIP traces: > > incoming call: http://pastebin.com/QK9K3KL9 > outgoing call: http://pastebin.com/zzxbm9mq > > There are call info headers in both traces. It's just the call-info headers > in the incoming trace look incorrect. > > I'm wondering if there has been any movement on this, or if I've just > screwed something up somewhere. > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp5191927p5191927.html > Sent from the freeswitch-users mailing list archive at Nabble.com. From chris.chen2004 at gmail.com Thu Jun 17 11:31:34 2010 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 17 Jun 2010 14:31:34 -0400 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> Message-ID: I second that Polycom SCA is working perfectly reliably with multiple accounts (if you have multiple lines such as Polycom IP 650). Thanks, Chris On Thu, Jun 17, 2010 at 2:12 PM, Brian West wrote: > The issue is the absence of the extra bits in the call-info header on every > packet involved in the call transaction... but its worse than that if you > want SCA you can only have ONE account on the phone and HOPE and pray it > behaves. I have tried to get Aastra to fix it but as far as I'm concerned > your best bet is to call Polycom and trade all those Aastra's for Polycom's > they have a trade up program and all you have to do is take a baseball bat > and make the Aastra's inoperable or not they seems to have the inoperable > part covered already right out of the box. > > /b > > On Jun 17, 2010, at 12:32 PM, Wes Pearce wrote: > > > > > Hello Users, > > > > I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in the > > office here. I've been trying to get SCA working unsuccessfully. While I > was > > digging around, I found this thread: > > > > > http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html > > > > Which describes my exact problem. SCA works neatly on outbound calls, and > > SCA works on incoming calls until the phone is answered... at which point > > the call dissapears from the other lines. > > > > Brian West mentioned this has something to do with Aastras inconsistent > > call-info headers. > > > > I've been speaking with a support representative from Aastra named Brian > > Epps, who says their engineers are unaware of any SCA problems. > > > > Here are the appropriate SIP traces: > > > > incoming call: http://pastebin.com/QK9K3KL9 > > outgoing call: http://pastebin.com/zzxbm9mq > > > > There are call info headers in both traces. It's just the call-info > headers > > in the incoming trace look incorrect. > > > > I'm wondering if there has been any movement on this, or if I've just > > screwed something up somewhere. > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp5191927p5191927.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/45b287fd/attachment.html From brian at freeswitch.org Thu Jun 17 11:40:14 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Jun 2010 13:40:14 -0500 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> Message-ID: <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> The Linksys/CiscoSPA's do too :P And those interop between each other correctly. /b On Jun 17, 2010, at 1:31 PM, Chris Chen wrote: > I second that Polycom SCA is working perfectly reliably with multiple accounts (if you have multiple lines such as Polycom IP 650). > Thanks, > Chris > > On Thu, Jun 17, 2010 at 2:12 PM, Brian West wrote: > The issue is the absence of the extra bits in the call-info header on every packet involved in the call transaction... but its worse than that if you want SCA you can only have ONE account on the phone and HOPE and pray it behaves. I have tried to get Aastra to fix it but as far as I'm concerned your best bet is to call Polycom and trade all those Aastra's for Polycom's they have a trade up program and all you have to do is take a baseball bat and make the Aastra's inoperable or not they seems to have the inoperable part covered already right out of the box. > > /b From testeador01 at gmail.com Thu Jun 17 12:16:26 2010 From: testeador01 at gmail.com (Milena) Date: Thu, 17 Jun 2010 14:16:26 -0500 Subject: [Freeswitch-users] Issue with an IVR-menu In-Reply-To: References: Message-ID: The problem was the empty digits="" that was added when doing tests and the menu hadn't been reloaded until i restarted freeswitch >_< thank you anyways :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/a9f3d36e/attachment.html From wespear at gmail.com Thu Jun 17 12:43:16 2010 From: wespear at gmail.com (Wes Pearce) Date: Thu, 17 Jun 2010 12:43:16 -0700 (PDT) Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> Message-ID: <1276803796935-5192528.post@n2.nabble.com> Lame lame lame! Thanks though guys, you rock. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp5191927p5192528.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Jun 17 12:49:04 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Jun 2010 14:49:04 -0500 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: <1276803796935-5192528.post@n2.nabble.com> References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> <1276803796935-5192528.post@n2.nabble.com> Message-ID: <9BF1AD4C-738E-4788-BF46-B85BB5EB0520@freeswitch.org> /me falls over laughing. I said just about the same thing but a few expletives inter weaved with it. /b On Jun 17, 2010, at 2:43 PM, Wes Pearce wrote: > > Lame lame lame! > > Thanks though guys, you rock. From anthony.minessale at gmail.com Thu Jun 17 12:54:22 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Jun 2010 14:54:22 -0500 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: <1276803796935-5192528.post@n2.nabble.com> References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> <1276803796935-5192528.post@n2.nabble.com> Message-ID: I just pinged them again, it's been 6 months since we reported the problem. I don't know what else to say besides they don't care. On Thu, Jun 17, 2010 at 2:43 PM, Wes Pearce wrote: > > Lame lame lame! > > Thanks though guys, you rock. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp5191927p5192528.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100617/56d8ef5c/attachment.html From mark.maly at molcs.org Thu Jun 17 14:04:01 2010 From: mark.maly at molcs.org (Mark Maly) Date: Thu, 17 Jun 2010 16:04:01 -0500 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> <1276803796935-5192528.post@n2.nabble.com> Message-ID: <00d701cb0e60$9cf23780$d6d6a680$@maly@molcs.org> All, Not smart enough to check to the "bit level" but received this from Aastra Tech support earlier and posted to another thread here, but . "Your ticket number is xxxxx. We have just released a new GA firmware that I would like for you to load onto one of your phones to see if it resolves the issue. You can download this from our website at www.aastratelecom.com/support and click on Download Area and select the 6731i. The firmware version you are looking for is 2.6 and it is listed under Current Software Release. Please load that on a test phone and let us know the results. Thank you, Jessie Fetter Aastra Customer Technical Support support at aastra.com www.aastratelecom.com/support 800-574-1611 " It appears this fixed the problems I was encountering, but I'm not comfortable saying it'll always work. I don't really have a valid test plan, but SCA "appears" to work with the new firmware.. Mark From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, June 17, 2010 2:54 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? I just pinged them again, it's been 6 months since we reported the problem. I don't know what else to say besides they don't care. On Thu, Jun 17, 2010 at 2:43 PM, Wes Pearce wrote: Lame lame lame! Thanks though guys, you rock. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp 5191927p5192528.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20100617/e6ceeb3d/attachment.html From brian at freeswitch.org Thu Jun 17 14:11:19 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Jun 2010 16:11:19 -0500 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: <00d701cb0e60$9cf23780$d6d6a680$@maly@molcs.org> References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> <1276803796935-5192528.post@n2.nabble.com> <00d701cb0e60$9cf23780$d6d6a680$@maly@molcs.org> Message-ID: <7CFA8A39-97EB-4D46-986C-6891C555F351@freeswitch.org> No it doesn't fully fix it.. again the phone DOES IT WRONG. We hacked FreeSWITCH to guess about it. Its still going to mess up. /b On Jun 17, 2010, at 4:04 PM, Mark Maly wrote: > All, > > Not smart enough to check to the ?bit level? but received this from Aastra Tech support earlier and posted to another thread here, but ? > > ?Your ticket number is xxxxx. > > We have just released a new GA firmware that I would like for you to load onto one of your phones to see if it resolves the issue. You can download this from our website at www.aastratelecom.com/support and click on Download Area and select the 6731i. The firmware version you are looking for is 2.6 and it is listed under Current Software Release. > > Please load that on a test phone and let us know the results. > > Thank you, > > Jessie Fetter > > Aastra Customer Technical Support > support at aastra.com > www.aastratelecom.com/support > 800-574-1611 > ? > > It appears this fixed the problems I was encountering, but I?m not comfortable saying it?ll always work. I don?t really have a valid test plan, but SCA ?appears? to work with the new firmware?. > > Mark From mark.maly at molcs.org Thu Jun 17 14:16:24 2010 From: mark.maly at molcs.org (Mark Maly) Date: Thu, 17 Jun 2010 16:16:24 -0500 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: <7CFA8A39-97EB-4D46-986C-6891C555F351@freeswitch.org> References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> <1276803796935-5192528.post@n2.nabble.com> <00d701cb0e60$9cf23780$d6d6a680$@maly@molcs.org> <7CFA8A39-97EB-4D46-986C-6891C555F351@freeswitch.org> Message-ID: <00e201cb0e62$581dc140$085943c0$@maly@molcs.org> Thanks... Mark -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 17, 2010 4:11 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? No it doesn't fully fix it.. again the phone DOES IT WRONG. We hacked FreeSWITCH to guess about it. Its still going to mess up. /b On Jun 17, 2010, at 4:04 PM, Mark Maly wrote: > All, > > Not smart enough to check to the "bit level" but received this from Aastra Tech support earlier and posted to another thread here, but . > > "Your ticket number is xxxxx. > > We have just released a new GA firmware that I would like for you to load onto one of your phones to see if it resolves the issue. You can download this from our website at www.aastratelecom.com/support and click on Download Area and select the 6731i. The firmware version you are looking for is 2.6 and it is listed under Current Software Release. > > Please load that on a test phone and let us know the results. > > Thank you, > > Jessie Fetter > > Aastra Customer Technical Support > support at aastra.com > www.aastratelecom.com/support > 800-574-1611 > " > > It appears this fixed the problems I was encountering, but I'm not comfortable saying it'll always work. I don't really have a valid test plan, but SCA "appears" to work with the new firmware.. > > Mark _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 dcolombo at voismart.it Thu Jun 17 14:20:18 2010 From: dcolombo at voismart.it (Davide Colombo) Date: Thu, 17 Jun 2010 23:20:18 +0200 (CEST) Subject: [Freeswitch-users] Question about execute_on_answer In-Reply-To: <1023238496.3886.1276809411075.JavaMail.root@mx.voismart.com> Message-ID: <974900369.3888.1276809618010.JavaMail.root@mx.voismart.com> Hi, first of all, thanks for your support! I followed Steven's advice, using a script with exec in answer confirm to play a file and to read dtmf. In this way i'm able to connect caller party with called party, hang-up caller party, move to voicemail caller party, but with multiple calls, phones that don't answer, they continue to ring. To avoid this behaviour i tried in a different way. Calls start with a multiple originate: originate user/1001,user/1002,user/1003 exten-to-ivr When one user answers, he is transfered to an ivr where he can choose the action. In this way caller party remains in ringing state and other phones stop to ring. In exten-to-ivr caller party's uuid is passed to let connection with called party. Davide ----- Messaggio originale ----- Da: "Steven Ayre" A: freeswitch-users at lists.freeswitch.org Inviato: Marted?, 15 giugno 2010 11:06:37 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer Read the "exec in answer confirm" section of http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation I think you would be able to do what you're asking within a script that's executed this way, playing the file and reading DTMF then doing logic based on the pressed key from within the script. Any scripting language should work - so lua etc as well as javascript. -Steve On 15 June 2010 09:46, Davide Colombo < dcolombo at voismart.it > wrote: Hi, using group_confirm_file and group_confirm_key i have only one choice, connect called party with calling party (1 dtmf maps with group_confirm_key). What i'd like to do when called party answers, it's: press 1 to connect with calling party, press 2 to hangup calling party, press 3 move calling party to voicemail. Additionally, making parellel dial in bridge, called phones stop to ring only when one of them digits the group_confirm_key and not (my purpose) when one of them answers. These are the reasons to transfer called channel to an ivr context using "execute_on_answer" variable. Best Regards ----- Messaggio originale ----- Da: "Michael Collins" < msc at freeswitch.org > A: freeswitch-users at lists.freeswitch.org Inviato: Marted?, 15 giugno 2010 0:37:18 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer Aha! You are trying to do some sort of answer verification, correct? You do know that FS already has this feature, correct? Check it out: http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables Make sure that you aren't duplicating functionality before continuing. -MC On Mon, Jun 14, 2010 at 3:12 PM, Davide Colombo < dcolombo at voismart.it > wrote: Hi, when 1019 answers, i transfer its channel (called channel party) to an IVR context using "execute_on_answer". In this context the called party (1019 in this case) can listen to a message where he can choose to connect/hang-up the calling party using dtmf tones. In the meanwhile i need that calling party remains in a ringing state. Best regards ----- Messaggio originale ----- Da: "Michael Collins" < msc at freeswitch.org > A: freeswitch-users at lists.freeswitch.org Inviato: Luned?, 14 giugno 2010 23:46:42 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer I'm not sure I follow what you're trying to accomplish... You call from x1000 to x1019... and what exactly do you want to have happen when 1019 answers? -MC On Mon, Jun 14, 2010 at 8:22 AM, Davide Colombo < dcolombo at voismart.it > wrote: Hi all, i'm trying to use "execute_on_answer" channel variable with a transfer before a bridge: When i call 1019 from 1000 and 1019 answers, i have this situation: 1019 is transfered to myivrpresentation but in console i can see this error 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 sofia/internal/ 1000 at fs2-devel.voismart.net Restore previous codec PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] while 1000 goes to next action after bridge in the same extension. Is it possible to transfer called channel (when called party answers) to a particular extension and leave caller channel in ringing state? Best Regards _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 From casteven at gmail.com Thu Jun 17 14:33:57 2010 From: casteven at gmail.com (Campbell Steven) Date: Fri, 18 Jun 2010 09:33:57 +1200 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: <7CFA8A39-97EB-4D46-986C-6891C555F351@freeswitch.org> References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> <1276803796935-5192528.post@n2.nabble.com> <7CFA8A39-97EB-4D46-986C-6891C555F351@freeswitch.org> Message-ID: I'm quite keen to help out and try and document some of these cases in the wiki, if nothing else if we can provide documented issues with various handsets we have more chance for the owners of these handsets to get issues resolved with the vendor even if they aren't quite up to being able to debug the problem themselves with the vendor. It also means that people can make informed decisions when buying handsets. i.e. if you want working SCA then buy Polycom. I have access to a number of makes/models of handsets, but not Aastra unfortunately. Campbell On Fri, Jun 18, 2010 at 9:11 AM, Brian West wrote: > No it doesn't fully fix it.. again the phone DOES IT WRONG. ?We hacked FreeSWITCH to guess about it. ?Its still going to mess up. > > /b > > On Jun 17, 2010, at 4:04 PM, Mark Maly wrote: > >> All, >> >> Not smart enough to check to the ?bit level? but received this from Aastra Tech support earlier and posted to another thread here, but ? >> >> ?Your ticket number is xxxxx. >> >> We have just released a new GA firmware that I would like for you to load onto one of your phones to see if it resolves the issue. You can download this from our website at www.aastratelecom.com/support and click on Download Area and select the 6731i. The firmware version you are looking for is 2.6 and it is listed under Current Software Release. >> >> Please load that on a test phone and let us know the results. >> >> Thank you, >> >> Jessie Fetter >> >> Aastra Customer Technical Support >> support at aastra.com >> www.aastratelecom.com/support >> 800-574-1611 >> ? >> >> It appears this fixed the problems I was encountering, but I?m not comfortable saying it?ll always work. ?I don?t really have a valid test plan, but SCA ?appears? to work with the new firmware?. >> >> Mark > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 codeghar at gmail.com Thu Jun 17 15:31:31 2010 From: codeghar at gmail.com (Code Ghar) Date: Thu, 17 Jun 2010 17:31:31 -0500 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> <47EBBABB-F2AE-4128-854D-FDEC48020C51@freeswitch.org> Message-ID: I was reading up on some security advice and came across an article from Digium (http://blogs.digium.com/2009/03/28/sip-security/). A few points that stood out for me were: "Make your SIP usernames different than your extensions". This sounds like good advice because now an attacker has to guess the user name and password instead of just a password. The biggest benefit to this is that even if someone knows the format of your extension numbers, they are not able to use it for registration credentials. Of course, the issue of DoS using a large number of simultaneous authentication requests still remains. How can we deal with this? "... reject bad authentication requests on valid usernames with the same rejection information as with invalid usernames ..." How can we do this in FS? I haven't looked at this yet so maybe FS does such a thing anyways; advice from your experience and expertise required. "Allow only one or two calls at a time per SIP entity, where possible." This is standard practice in prepaid phone card business. They allow only one simultaneous call and if you try to use the PIN to make another call it's not allowed. At the application level we may be able to do this in FS but it reminds me of another scenario. Let's say a user's registration is compromised. Can we prevent registration with same credentials from two different locations in FS? For example, one registration attempt is from New York and the other from Baltimore. This would affect the legitimate user because they might not be able to register. Does anyone restrict this way using FS? Apart from the article mentioned, there are some other issues, one of which is ANI authentication. A lot of pinless prepaid phone services use your caller ID to authenticate and authorize a call. you give them your phone number and when you call from there you can use the service. With VoIP it's very easy to spoof ANI. I believe the concept of "context" helps a lot here. Say you know the ANI's you assigned to your registered users. They would be in the "default" context, for example. Now if an attacker uses the same ANI but tries not from a registered endpoint but say from one of your carriers. His context would be "public", for example. This way the combination of ANI with context can prevent such attacks. But what if the authorized ANI (say 4145550000) is not from a registered endpoint but from a cell phone, similar to the pinless services offered today. In this scenario I think there's not much we can do. If we have given a user an access number (say 2125550000) provided to us by a DID provider, say ABC, then whenever someone dials 2125550000 then we get the call only through ABC, whom we have allowed. But we can't be sure that 4145550000 is actually 4145550000 and not someone pretending to be them. Has anyone figured out a way to detect and/or prevent such fraudulent calls? Is it even possible? On Tue, Jun 15, 2010 at 4:59 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > We have some parameters to disable some commonly abused features in a > carrier env. > > in the sofia profile > > set manage-presence to false > > If you are not providing registration or client transfers disable them > completely > set disable-transfer to true > set disable-register to true > > > And register for ClueCon where someone will probably give a talk on VoIP > security! > > > -- > 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/20100617/12c33a84/attachment.html From codeghar at gmail.com Thu Jun 17 15:35:25 2010 From: codeghar at gmail.com (Code Ghar) Date: Thu, 17 Jun 2010 17:35:25 -0500 Subject: [Freeswitch-users] default SIP registration timeout In-Reply-To: References: Message-ID: What kind of registration? Are we talking about an endpoint, such as ATA or softphone, or a gateway? On Mon, Jun 14, 2010 at 2:09 AM, Madovsky wrote: > Is there any xml param for SIP registration timeout ? > > Thanks > > F > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/f4eda773/attachment.html From sean at obscuradigital.com Thu Jun 17 15:38:02 2010 From: sean at obscuradigital.com (Sean Holt) Date: Thu, 17 Jun 2010 15:38:02 -0700 Subject: [Freeswitch-users] 4 second delay In-Reply-To: Message-ID: Hey MC, Did you by chance get an opportunity to look over my configuration in the pastebin? Thanks Sean On 6/15/10 12:45 PM, "Michael Collins" wrote: > Pastebin, please. > > On Tue, Jun 15, 2010 at 12:19 PM, Sean Holt wrote: >> Should I put the freeswitch log into the pastebin or attach it? >> >> >> On 6/15/10 11:11 AM, "Michael Collins" > > wrote: >> >>> Sure. Put all this information into pastebin.freeswitch.org >>> : >>> >>> >>> Your topology, including the SIP provider, any routers/firewalls (including >>> make & model), phone make & model >>> Your relevant dialplan >>> Console log of the call, with SIP trace turned on >>> >>> You can also email me the .pcap file, but I don't want to review it until >>> you've got all the above information posted so that we can get a good >>> picture of what's happening... >>> >>> Thanks! >>> -MC >>> >>> >>> On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt >> > wrote: >>>> Hey MC, would you be willing to look over my tcpdump to help determine >>>> where the failure is happening? >>>> >>>> Sean >>>> >>>> >>>> >>>> On 6/14/10 12:32 PM, "Michael Collins" >>> > wrote: >>>> >>>>> Hop on the console, turn on siptrace and watch the call flow for clues. >>>>> You might need to do a tcpdump capturing both signaling and media and then >>>>> analyze in Wireshark to see what exactly is happening. >>>>> -MC >>>>> >>>>> On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt >>>> > >>>>> wrote: >>>>>> Hello list, >>>>>> >>>>>> I?ve been dealing with a particular issue with in-coming calls. ? >>>>>> Leg A calls into the office, then Leg B (endpoint) picks up call but >>>>>> hears nothing on other side. ?Wait 4 sec call completes and Leg B can >>>>>> hear the other person. ? >>>>>> >>>>>> I have Centos 5.4 >>>>>> Latest git build >>>>>> Polycom phones >>>>>> >>>>>> Calling out is not a problem. >>>>>> >>>>>> Not sure how to troubleshoot this issue or maybe there?s a delay setting >>>>>> in the sip profile that waits for the channel to complete. >>>>>> >>>>>> Thanks for the help >>>>>> Sean >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.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 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100617/70dba90a/attachment-0001.html From codeghar at gmail.com Thu Jun 17 15:46:56 2010 From: codeghar at gmail.com (Code Ghar) Date: Thu, 17 Jun 2010 17:46:56 -0500 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: References: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> Message-ID: I haven't had a chance to actually test this config in lab but I will in the coming days. One issue I see is when we use registered endpoints, such as ATAs, behind NAT: bypass_media fails in this case because FS uses the private network IP instead of the public network IP in SDP. In case of a large number of such registered endpoints, we may have to make them talk with an FS-RTP server directly because they don't use bypass_media while FS-SIP does. Can we use bypass_media_after_reinvite for them or would it have the same behavior of using private IP? Or we could use proxy_media; which is not want we really want to do but could work in a pinch. Or is there a better way to combine bypass_media with NAT'ed endpoints? On Mon, Jun 7, 2010 at 6:29 AM, David Ponzone wrote: > Mike, > > You're right, it can be achieved with SIP now that I think a bit more about > it. > The idea was to allow adding multiple media gateways when required, so the > media gateways should not be facing the carriers as some of them do > SIP-filtering, but should only be advertised in the SDP. > > So SIP-only boxes (doing bypass-media) should face the carriers to handle > the trunking. > In the middle, we would then have the media gateways, doing SIP and mostly > RTP. > But I guess we dont want customers to register and to send calls to a media > gateway, so we need another set of SIP boxes on the other side, doing > bypass-media also. > > So it would like this: > > ------sip-----FS-RTP-1-----sip------ > FS-SIP-Internal-1 > ------sip-----FS-RTP-2-----sip------FS-SIP-External-1----sip-----Carriers > ------sip-----FS-RTP-3-----sip------ > FS-SIP-Internal-2 > -------sip----FS-RTP-4-----sip------FS-SIP-External-2-----sip----Carriers > -------sip----FS-RTP-5-----sip------ > > Thanks to bypass-media, the RTP streams would go from customer to FS-RTP-x > to Carriers, and reverse. > And I don't see any reason why the same set of FS-SIP boxes could not be > used for both internal and external borders. > > Is there something wrong in this ? > > Code, does it help ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 05/06/2010 ? 19:54, Michael Jerris a ?crit : > > Why would it be an advantage to have your media proxies use another > protocol? > > Mike > > On Jun 4, 2010, at 1:59 AM, David Ponzone wrote: > > It doesn't solve the issue that all the media servers will do signaling > too, and will talk SIP with the carriers. > So the carriers will need to allow all the media servers . > > The only clean solution to avoid that, I think, is to have signaling boxes > allocating resources from media servers with another protocol than SIP. > RTPproxy does that I think, but I am not sure how it works. > > David Ponzone > > _______________________________________________ > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100617/06fc3b95/attachment.html From msc at freeswitch.org Thu Jun 17 15:50:02 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Jun 2010 15:50:02 -0700 Subject: [Freeswitch-users] 4 second delay In-Reply-To: References: Message-ID: On Thu, Jun 17, 2010 at 3:38 PM, Sean Holt wrote: > Hey MC, > Did you by chance get an opportunity to look over my configuration in the > pastebin? > > Thanks > Sean > > Yes. Question - are you deliberately calling four or five different phones when this call comes in? Or do you hve SCA/SLA set up? It looks like a bunch of phones all get rung when this call comes in. Is that what you are trying to do? To make sure there isn't an issue with SCA try sending a call to a phone that is not sharing any of its lines and capture the same information. See if the 4 sec lag is there also. -MC > > On 6/15/10 12:45 PM, "Michael Collins" wrote: > > Pastebin, please. > > On Tue, Jun 15, 2010 at 12:19 PM, Sean Holt > wrote: > > Should I put the freeswitch log into the pastebin or attach it? > > > On 6/15/10 11:11 AM, "Michael Collins" http://msc at freeswitch.org> > wrote: > > Sure. Put all this information into pastebin.freeswitch.org < > http://pastebin.freeswitch.org> : > > > > Your topology, including the SIP provider, any routers/firewalls (including > make & model), phone make & model > Your relevant dialplan > Console log of the call, with SIP trace turned on > > You can also email me the .pcap file, but I don't want to review it until > you've got all the above information posted so that we can get a good > picture of what's happening... > > Thanks! > -MC > > > On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt http://sean at obscuradigital.com> > wrote: > > Hey MC, would you be willing to look over my tcpdump to help determine > where the failure is happening? > > Sean > > > > On 6/14/10 12:32 PM, "Michael Collins" http://msc at freeswitch.org> > wrote: > > Hop on the console, turn on siptrace and watch the call flow for clues. You > might need to do a tcpdump capturing both signaling and media and then > analyze in Wireshark to see what exactly is happening. > -MC > > On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt http://sean at obscuradigital.com> > wrote: > > Hello list, > > I?ve been dealing with a particular issue with in-coming calls. > Leg A calls into the office, then Leg B (endpoint) picks up call but hears > nothing on other side. Wait 4 sec call completes and Leg B can hear the > other person. > > I have Centos 5.4 > Latest git build > Polycom phones > > Calling out is not a problem. > > Not sure how to troubleshoot this issue or maybe there?s a delay setting in > the sip profile that waits for the channel to complete. > > Thanks for the help > Sean > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > http://FreeSWITCH-users at lists.freeswitch.org> < > http://FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > http://FreeSWITCH-users at lists.freeswitch.org> < > http://FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > http://FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > http://FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > > > ------------------------------ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100617/4ab28dc8/attachment-0001.html From brian at freeswitch.org Thu Jun 17 15:53:15 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Jun 2010 17:53:15 -0500 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> <47EBBABB-F2AE-4128-854D-FDEC48020C51@freeswitch.org> Message-ID: <59D37181-66F1-4070-AB75-4A2D51842ABA@freeswitch.org> You really should try FreeSWICH and understand the security. In the end its YOUR job to secure it... not ours. I can't stop someone from setting up FreeSWITCH in an insecure way. Just like you can't stop someone from setting up a CGI script that owns your box. On Jun 17, 2010, at 5:31 PM, Code Ghar wrote: > I was reading up on some security advice and came across an article from Digium (http://blogs.digium.com/2009/03/28/sip-security/). A few points that stood out for me were: > > "Make your SIP usernames different than your extensions". This sounds like good advice because now an attacker has to guess the user name and password instead of just a password. The biggest benefit to this is that even if someone knows the format of your extension numbers, they are not able to use it for registration credentials. Of course, the issue of DoS using a large number of simultaneous authentication requests still remains. How can we deal with this? How about you set the AUTH username different then the username and then have a password too? > > "... reject bad authentication requests on valid usernames with the same rejection information as with invalid usernames ..." How can we do this in FS? I haven't looked at this yet so maybe FS does such a thing anyways; advice from your experience and expertise required. All bad requests get a 403. All requests good or bad get a 401 then a 403 if invalid. No indications at all that the username is valid. > "Allow only one or two calls at a time per SIP entity, where possible." This is standard practice in prepaid phone card business. They allow only one simultaneous call and if you try to use the PIN to make another call it's not allowed. At the application level we may be able to do this in FS but it reminds me of another scenario. Let's say a user's registration is compromised. Can we prevent registration with same credentials from two different locations in FS? For example, one registration attempt is from New York and the other from Baltimore. This would affect the legitimate user because they might not be able to register. Does anyone restrict this way using FS? Mod_limit. > Apart from the article mentioned, there are some other issues, one of which is ANI authentication. A lot of pinless prepaid phone services use your caller ID to authenticate and authorize a call. you give them your phone number and when you call from there you can use the service. With VoIP it's very easy to spoof ANI. I believe the concept of "context" helps a lot here. Say you know the ANI's you assigned to your registered users. They would be in the "default" context, for example. Now if an attacker uses the same ANI but tries not from a registered endpoint but say from one of your carriers. His context would be "public", for example. This way the combination of ANI with context can prevent such attacks. Never a good idea to use caller id as auth. > But what if the authorized ANI (say 4145550000) is not from a registered endpoint but from a cell phone, similar to the pinless services offered today. In this scenario I think there's not much we can do. If we have given a user an access number (say 2125550000) provided to us by a DID provider, say ABC, then whenever someone dials 2125550000 then we get the call only through ABC, whom we have allowed. But we can't be sure that 4145550000 is actually 4145550000 and not someone pretending to be them. Has anyone figured out a way to detect and/or prevent such fraudulent calls? Is it even possible? NO. /b From msc at freeswitch.org Thu Jun 17 16:02:24 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Jun 2010 16:02:24 -0700 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> <47EBBABB-F2AE-4128-854D-FDEC48020C51@freeswitch.org> Message-ID: On Thu, Jun 17, 2010 at 3:31 PM, Code Ghar wrote: > I was reading up on some security advice and came across an article from > Digium (http://blogs.digium.com/2009/03/28/sip-security/). A few points > that stood out for me were: > > "Make your SIP usernames different than your extensions". This sounds like > good advice because now an attacker has to guess the user name and password > instead of just a password. The biggest benefit to this is that even if > someone knows the format of your extension numbers, they are not able to use > it for registration credentials. Of course, the issue of DoS using a large > number of simultaneous authentication requests still remains. How can we > deal with this? > Use fail2ban or some other mechanism. If someone is sending mass numbers of SUBSCRIBEs then you should be handling that the same way you handle all brute-force attacks - at the firewall. Although you should definitely not make hacking your VoIP system easy, you should also keep in mind that external security is an absolute must. Use the layered approach > > "... reject bad authentication requests on valid usernames with the same > rejection information as with invalid usernames ..." How can we do this in > FS? I haven't looked at this yet so maybe FS does such a thing anyways; > advice from your experience and expertise required. > This I don't know. Is it handled down in Sofia? Brian or Tony could tell us. > > "Allow only one or two calls at a time per SIP entity, where possible." > This is standard practice in prepaid phone card business. They allow only > one simultaneous call and if you try to use the PIN to make another call > it's not allowed. At the application level we may be able to do this in FS > but it reminds me of another scenario. Let's say a user's registration is > compromised. Can we prevent registration with same credentials from two > different locations in FS? For example, one registration attempt is from New > York and the other from Baltimore. This would affect the legitimate user > because they might not be able to register. Does anyone restrict this way > using FS? > I'm sure you could use a combination of ACLs and mod_limit. > > Apart from the article mentioned, there are some other issues, one of which > is ANI authentication. A lot of pinless prepaid phone services use your > caller ID to authenticate and authorize a call. you give them your phone > number and when you call from there you can use the service. With VoIP it's > very easy to spoof ANI. I believe the concept of "context" helps a lot here. > Say you know the ANI's you assigned to your registered users. They would be > in the "default" context, for example. Now if an attacker uses the same ANI > but tries not from a registered endpoint but say from one of your carriers. > His context would be "public", for example. This way the combination of ANI > with context can prevent such attacks. > I cringe at the thought of doing *ANY* kind of authentication based on incoming caller ID... > > But what if the authorized ANI (say 4145550000) is not from a registered > endpoint but from a cell phone, similar to the pinless services offered > today. In this scenario I think there's not much we can do. If we have given > a user an access number (say 2125550000) provided to us by a DID provider, > say ABC, then whenever someone dials 2125550000 then we get the call only > through ABC, whom we have allowed. But we can't be sure that 4145550000 is > actually 4145550000 and not someone pretending to be them. Has anyone > figured out a way to detect and/or prevent such fraudulent calls? Is it even > possible? > I think this is where mod_cleo comes in. :P Seriously, you probably should treat inbound calls as untrusted regardless of where they're from, at least until you've done some level of verification. A PIN code isn't hyper-secure, but it is better than nothing. You can't prevent all of these attacks. The best you can hope to do is to make the attacks as difficult as reasonably possible without irritating your paying customers. I invite others to chime in. Also, I've started an informal security best practices page on the wiki: http://wiki.freeswitch.org/wiki/Security_bp Everyone put your thoughts up and we'll formalize it at a later time. -MC > > > > On Tue, Jun 15, 2010 at 4:59 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> We have some parameters to disable some commonly abused features in a >> carrier env. >> >> in the sofia profile >> >> set manage-presence to false >> >> If you are not providing registration or client transfers disable them >> completely >> set disable-transfer to true >> set disable-register to true >> >> >> And register for ClueCon where someone will probably give a talk on VoIP >> security! >> >> >> -- >> 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 >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/7a2fe1da/attachment.html From msc at freeswitch.org Thu Jun 17 16:05:14 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Jun 2010 16:05:14 -0700 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: <59D37181-66F1-4070-AB75-4A2D51842ABA@freeswitch.org> References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> <47EBBABB-F2AE-4128-854D-FDEC48020C51@freeswitch.org> <59D37181-66F1-4070-AB75-4A2D51842ABA@freeswitch.org> Message-ID: On Thu, Jun 17, 2010 at 3:53 PM, Brian West wrote: > You really should try FreeSWICH and understand the security. In the end > its YOUR job to secure it... not ours. I can't stop someone from setting up > FreeSWITCH in an insecure way. Just like you can't stop someone from > setting up a CGI script that owns your box. > > On Jun 17, 2010, at 5:31 PM, Code Ghar wrote: > > > I was reading up on some security advice and came across an article from > Digium (http://blogs.digium.com/2009/03/28/sip-security/). A few points > that stood out for me were: > > > > "Make your SIP usernames different than your extensions". This sounds > like good advice because now an attacker has to guess the user name and > password instead of just a password. The biggest benefit to this is that > even if someone knows the format of your extension numbers, they are not > able to use it for registration credentials. Of course, the issue of DoS > using a large number of simultaneous authentication requests still remains. > How can we deal with this? > > How about you set the AUTH username different then the username and then > have a password too? > FYI, there are soft phones out there that FAIL at this: wxCommunicator GoldMine CRM Softphone (Front Range Solutions) These two phones do not support the concept of "auth username" - they ASSUME that the username is the auth username. Naughty naughty. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/13c6efab/attachment.html From sos at sokhapkin.dyndns.org Thu Jun 17 16:13:04 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Thu, 17 Jun 2010 19:13:04 -0400 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: <59D37181-66F1-4070-AB75-4A2D51842ABA@freeswitch.org> References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> <59D37181-66F1-4070-AB75-4A2D51842ABA@freeswitch.org> Message-ID: <201006171913.04207.sos@sokhapkin.dyndns.org> The security related problems were discussed on almost every soft switch mailing lists I'm watching. And the outcome was (in plain English which Americans do not understand) - never provide working configuration samples in the softswitch distribution :-( Let the end user build the configuration. On Thursday 17 June 2010, Brian West wrote: > You really should try FreeSWICH and understand the security. In the end > its YOUR job to secure it... not ours. I can't stop someone from setting > up FreeSWITCH in an insecure way. Just like you can't stop someone from > setting up a CGI script that owns your box. > > On Jun 17, 2010, at 5:31 PM, Code Ghar wrote: > > I was reading up on some security advice and came across an article from > > Digium (http://blogs.digium.com/2009/03/28/sip-security/). A few points > > that stood out for me were: > > > > "Make your SIP usernames different than your extensions". This sounds > > like good advice because now an attacker has to guess the user name and > > password instead of just a password. The biggest benefit to this is that > > even if someone knows the format of your extension numbers, they are not > > able to use it for registration credentials. Of course, the issue of DoS > > using a large number of simultaneous authentication requests still > > remains. How can we deal with this? > > How about you set the AUTH username different then the username and then > have a password too? > > > "... reject bad authentication requests on valid usernames with the same > > rejection information as with invalid usernames ..." How can we do this > > in FS? I haven't looked at this yet so maybe FS does such a thing > > anyways; advice from your experience and expertise required. > > All bad requests get a 403. All requests good or bad get a 401 then a 403 > if invalid. No indications at all that the username is valid. > > > "Allow only one or two calls at a time per SIP entity, where possible." > > This is standard practice in prepaid phone card business. They allow only > > one simultaneous call and if you try to use the PIN to make another call > > it's not allowed. At the application level we may be able to do this in > > FS but it reminds me of another scenario. Let's say a user's registration > > is compromised. Can we prevent registration with same credentials from > > two different locations in FS? For example, one registration attempt is > > from New York and the other from Baltimore. This would affect the > > legitimate user because they might not be able to register. Does anyone > > restrict this way using FS? > > Mod_limit. > > > Apart from the article mentioned, there are some other issues, one of > > which is ANI authentication. A lot of pinless prepaid phone services use > > your caller ID to authenticate and authorize a call. you give them your > > phone number and when you call from there you can use the service. With > > VoIP it's very easy to spoof ANI. I believe the concept of "context" > > helps a lot here. Say you know the ANI's you assigned to your registered > > users. They would be in the "default" context, for example. Now if an > > attacker uses the same ANI but tries not from a registered endpoint but > > say from one of your carriers. His context would be "public", for > > example. This way the combination of ANI with context can prevent such > > attacks. > > Never a good idea to use caller id as auth. > > > But what if the authorized ANI (say 4145550000) is not from a registered > > endpoint but from a cell phone, similar to the pinless services offered > > today. In this scenario I think there's not much we can do. If we have > > given a user an access number (say 2125550000) provided to us by a DID > > provider, say ABC, then whenever someone dials 2125550000 then we get the > > call only through ABC, whom we have allowed. But we can't be sure that > > 4145550000 is actually 4145550000 and not someone pretending to be them. > > Has anyone figured out a way to detect and/or prevent such fraudulent > > calls? Is it even possible? > > NO. > > /b > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Thu Jun 17 16:21:27 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Jun 2010 16:21:27 -0700 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: <201006171913.04207.sos@sokhapkin.dyndns.org> References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> <59D37181-66F1-4070-AB75-4A2D51842ABA@freeswitch.org> <201006171913.04207.sos@sokhapkin.dyndns.org> Message-ID: On Thu, Jun 17, 2010 at 4:13 PM, Sergey Okhapkin wrote: > The security related problems were discussed on almost every soft switch > mailing lists I'm watching. And the outcome was (in plain English which > Americans do not understand) - never provide working configuration samples > in > the softswitch distribution :-( Let the end user build the configuration. > I don't know that I agree with such an extreme view. Better to give them an example of a locked-down config than to let them flounder on their own and possibly make something horrible. That being said, anyone using FS in the enterprise *SHOULD* consult a security professional to make sure that everything on their LAN/WAN is up to snuff. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/0f4c6049/attachment.html From mranga at gmail.com Thu Jun 17 16:23:53 2010 From: mranga at gmail.com (M. Ranganathan) Date: Thu, 17 Jun 2010 19:23:53 -0400 Subject: [Freeswitch-users] Playing a voice prompt in a conference Message-ID: Hello, I want to play a prompt ( wav file ) in a conference after two participants join it. What is the best way to accomplish something like that in FreeSWITCH? Are there any examples I can look at? Thank you in advance for any help Ranga -- M. Ranganathan From sos at sokhapkin.dyndns.org Thu Jun 17 17:11:25 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Thu, 17 Jun 2010 20:11:25 -0400 Subject: [Freeswitch-users] Your Security Best Practices In-Reply-To: References: <2f42aded9f4727bdf042877138fc9d0f@mail.gmail.com> <201006171913.04207.sos@sokhapkin.dyndns.org> Message-ID: <201006172011.26012.sos@sokhapkin.dyndns.org> Give them a non-working example. And provide a comment on almost every line why this will not work. Let the customers exercise their brain first. You don't need dumb customers, am I right? You'll see a good outcome. Sorry, but I can't resist. It's my daily work to get rid of unwanted customers. "Unwanted Customers" doesn't sound good in any language. But it's a reality... On Thursday 17 June 2010, Michael Collins wrote: > On Thu, Jun 17, 2010 at 4:13 PM, Sergey Okhapkin > > wrote: > > The security related problems were discussed on almost every soft switch > > mailing lists I'm watching. And the outcome was (in plain English which > > Americans do not understand) - never provide working configuration > > samples in > > the softswitch distribution :-( Let the end user build the configuration. > > I don't know that I agree with such an extreme view. Better to give them an > example of a locked-down config than to let them flounder on their own and > possibly make something horrible. That being said, anyone using FS in the > enterprise *SHOULD* consult a security professional to make sure that > everything on their LAN/WAN is up to snuff. > > -MC > From jmesquita at freeswitch.org Thu Jun 17 17:37:19 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 17 Jun 2010 21:37:19 -0300 Subject: [Freeswitch-users] Playing a voice prompt in a conference In-Reply-To: References: Message-ID: I can think of more then a couple of ways to accomplish this. Would you mind filling a little detail of where exactly you want this? On dialplan? With ESL? On script? Regards, Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 On Thu, Jun 17, 2010 at 8:23 PM, M. Ranganathan wrote: > Hello, > > I want to play a prompt ( wav file ) in a conference after two > participants join it. What is the best way to accomplish something > like that in FreeSWITCH? Are there any examples I can look at? > > Thank you in advance for any help > > Ranga > > -- > M. Ranganathan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100617/1aac6a57/attachment.html From sean at obscuradigital.com Thu Jun 17 18:03:09 2010 From: sean at obscuradigital.com (Sean Holt) Date: Thu, 17 Jun 2010 18:03:09 -0700 Subject: [Freeswitch-users] 4 second delay In-Reply-To: Message-ID: I am deliberately calling 5 phones, in a ring group. Previously I did try a single phone, but same issue. I don?t have shared line enables. Sean On 6/17/10 3:50 PM, "Michael Collins" wrote: > > > On Thu, Jun 17, 2010 at 3:38 PM, Sean Holt wrote: >> Hey MC, >> Did you by chance get an opportunity to look over my configuration in the >> pastebin? >> >> Thanks >> Sean >> > Yes. Question - are you deliberately calling four or five different phones > when this call comes in? Or do you hve SCA/SLA set up? It looks like a bunch > of phones all get rung when this call comes in. Is that what you are trying to > do? To make sure there isn't an issue with SCA try sending a call to a phone > that is not sharing any of its lines and capture the same information. See if > the 4 sec lag is there also. > -MC > >> >> >> On 6/15/10 12:45 PM, "Michael Collins" > > wrote: >> >>> Pastebin, please. >>> >>> On Tue, Jun 15, 2010 at 12:19 PM, Sean Holt >> > wrote: >>>> Should I put the freeswitch log into the pastebin or attach it? >>>> >>>> >>>> On 6/15/10 11:11 AM, "Michael Collins" >>> > wrote: >>>> >>>>> Sure. Put all this information into pastebin.freeswitch.org >>>>> >>>>> ? : >>>>> >>>>> >>>>> >>>>> Your topology, including the SIP provider, any routers/firewalls >>>>> (including make & model), phone make & model >>>>> Your relevant dialplan >>>>> Console log of the call, with SIP trace turned on >>>>> >>>>> You can also email me the .pcap file, but I don't want to review it until >>>>> you've got all the above information posted so that we can get a good >>>>> picture of what's happening... >>>>> >>>>> Thanks! >>>>> -MC >>>>> >>>>> >>>>> On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt >>>> > >>>>> wrote: >>>>>> Hey MC, would you be willing to look over my tcpdump to help determine >>>>>> where the failure is happening? >>>>>> >>>>>> Sean >>>>>> >>>>>> >>>>>> >>>>>> On 6/14/10 12:32 PM, "Michael Collins" >>>>> >>>>>> ? > wrote: >>>>>> >>>>>>> Hop on the console, turn on siptrace and watch the call flow for clues. >>>>>>> You might need to do a tcpdump capturing both signaling and media and >>>>>>> then analyze in Wireshark to see what exactly is happening. >>>>>>> -MC >>>>>>> >>>>>>> On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt >>>>>> >>>>>>> ? > wrote: Hello list, I?ve been dealing with a particular issue with in-coming calls. ? Leg A calls into the office, then Leg B (endpoint) picks up call but hears nothing on other side. ?Wait 4 sec call completes and Leg B can hear the other person. ? I have Centos 5.4 Latest git build Polycom phones Calling out is not a problem. Not sure how to troubleshoot this issue or maybe there?s a delay setting in the sip profile that waits for the channel to complete. Thanks for the help Sean _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.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 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100617/1fc42caf/attachment-0001.html From bandwidth.user at gmail.com Thu Jun 17 18:34:34 2010 From: bandwidth.user at gmail.com (roy) Date: Fri, 18 Jun 2010 09:34:34 +0800 Subject: [Freeswitch-users] mod_pocketsphinx In-Reply-To: References: <4C1997A0.1080909@gmail.com> Message-ID: <4C1ACD2A.70906@gmail.com> On Thursday, 17 June, 2010 08:34 PM, Brian West wrote: > your running ubuntu aren't you? FS on Debian/squeeze From brian at freeswitch.org Thu Jun 17 19:12:48 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Jun 2010 21:12:48 -0500 Subject: [Freeswitch-users] mod_pocketsphinx In-Reply-To: <4C1ACD2A.70906@gmail.com> References: <4C1997A0.1080909@gmail.com> <4C1ACD2A.70906@gmail.com> Message-ID: remove the pocketsphinx debs. /b On Jun 17, 2010, at 8:34 PM, roy wrote: > On Thursday, 17 June, 2010 08:34 PM, Brian West wrote: >> your running ubuntu aren't you? > > FS on Debian/squeeze > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 bandwidth.user at gmail.com Thu Jun 17 20:07:45 2010 From: bandwidth.user at gmail.com (roy) Date: Fri, 18 Jun 2010 11:07:45 +0800 Subject: [Freeswitch-users] mod_pocketsphinx In-Reply-To: References: <4C1997A0.1080909@gmail.com> <4C1ACD2A.70906@gmail.com> Message-ID: <4C1AE301.30405@gmail.com> Brian, On Friday, 18 June, 2010 10:12 AM, Brian West wrote: > remove the pocketsphinx debs. Thanks for the quick reply. I installed the whole thing using 'make current' with asr_tts/mod_pocketsphinx uncommented on modules.conf (if you're pointing to a .deb pocketsphinx install) From thangappan143 at gmail.com Thu Jun 17 21:56:58 2010 From: thangappan143 at gmail.com (Thangappan.M) Date: Fri, 18 Jun 2010 10:26:58 +0530 Subject: [Freeswitch-users] Build the mod_unimrcp module Message-ID: Dear all, I am in the process of developing IVR. So just planned to convert my application to handle the TTS voice engine which is supported by FreeSWITCH. Got the mod_unimrcp module which is used to recognize the speech and synthesize the text to voice. For building the mod_unimrcp modules done the following steps. Uncomment the mod_unimrcp line in the modules.conf file in the FreeSWITCH source Given make mod_unimrcp-install command. In modules.conf.xml uncomment the Configured the following dial plan --> While making the call to 4922 got the following error in the FreeSWITCH console. [INFO] mod_dialplan_xml.c:418 Processing thangappan->4922 in context default [NOTICE] mod_dptools.c:717 Channel [sofia/internal/1012 at 192.168.1.222] has been answered [INFO] mod_unimrcp.c:1499 speech_handle: name = unimrcp, rate = 8000, speed = 0, samples = 160, voice = , engine = unimrcp, param = nuance5-mrcp1 [INFO] mod_unimrcp.c:1502 voice = awb, rate = 8000 [NOTICE] mrcp_client.c:549 Create MRCP Handle 0x8b02400 [nuance5-mrcp1] [INFO] mrcp_client_session.c:142 Create Channel 0x8b02400 [INFO] mrcp_client_session.c:398 Receive App Request 0x8b02400 [2] [NOTICE] rtsp_client.c:255 Create RTSP Handle 0x8b04408 [INFO] mrcp_client.c:901 Add MRCP Handle 0x8b02400 [NOTICE] mrcp_client_session.c:718 Add Control Channel 0x8b02400 [INFO] mrcp_client_session.c:420 Send Offer 0x8b02400 [c:0 a:1 v:0] *[ERR] mod_unimrcp.c:965 (TTS-0) Timed out waiting for channel to be ready [ERR] switch_ivr_play_say.c:2104 Invalid TTS module!* [NOTICE] switch_core_state_machine.c:185 sofia/internal/1012 at 192.168.1.222 has executed the last dialplan instruction, hanging up. [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1012 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING] [NOTICE] switch_core_session.c:1179 Session 1 (sofia/internal/1012 at 192.168.1.222) Ended [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/1012 at 192.168.1.222 [CS_DESTROY] [NOTICE] switch_channel.c:669 New Channel sofia/internal/1012 at 192.168.1.222 [49ed5afa-79f0-11df-b531-3553f3a65c3c] So need to find a solution for that. -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/d16bbe90/attachment.html From lakindia89 at gmail.com Thu Jun 17 22:46:16 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Fri, 18 Jun 2010 11:16:16 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: Dear Antony, If leg timeout beats group_confirm_cancel_timeout then in wiki it is stated as follows: group_confirm_cancel_timeoutIf set, cancels a leg timeout after the call is answered.Can you please clarify me then, what is the use of group_confirm_cancel_timeout, if leg_timeout beats it. On Tue, Jun 15, 2010 at 10:53 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > leg timeout beats the group confirm timeouts > > > On Tue, Jun 15, 2010 at 12:28 AM, Nagalenoj H. wrote: > >> Dear friends, >> I've tried using the group_confirm_cancel_timeout channel variable. >> I've written a testing script to get digits before bridging. But, it doesn't >> seem to be working. >> >> My understanding after reading wiki is, >> * When I dial [leg_timeout=10]user/1005, if he answers before >> timeout and in the process of giving digits, then the call shouldn't be >> disconnected after the leg_timeout secs (10 sec in the example). >> >> But, When I experiment it, the call is getting disconnected after 10 >> seconds and it doesn't bother whether the callee has answered the >> call(Started giving digits) or not answered at all. >> >> I've checked it with nc as follows, >> >> sendmsg >> call-command: execute >> execute-app-name: set >> execute-app-arg: group_confirm_key=exec >> >> sendmsg >> call-command: execute >> execute-app-name: set >> execute-app-arg: group_confirm_file=perl /root/confirm.pl >> >> sendmsg >> call-command: execute >> execute-app-name: set >> execute-app-arg: group_confirm_cancel_timeout=1 >> >> sendmsg >> call-command: execute >> execute-app-name: bridge >> execute-app-arg: [leg_timeout=10]user/1005 >> >> And here is the script, >> >> use freeswitch; >> our $session; >> my $digit; >> >> while(1) { >> # Wait till response timeout for the first digit. >> $digit = $session->getDigits(1, "", 10000); >> freeswitch::consoleLog ("info","Digit>>".$digit."<<"); >> >> if (! $session->ready() ) { >> freeswitch::consoleLog("info","Going to Exit\n"); >> last; >> } >> if (defined $digit and $digit ne "" ) { >> freeswitch::consoleLog("info","DTMF received: $digit\n"); >> if ($digit eq '#') { >> return; >> } >> } >> else { >> freeswitch::consoleLog("info","Timeout\n"); >> $session->hangup(); >> } >> } >> 1; >> >> If my understanding is right then, I believe there is something wrong with >> channel_variable. >> >> Kindly help me to resolve this. >> >> -- >> Regards, >> Nagalenoj H. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/34059724/attachment.html From david.ponzone at gmail.com Thu Jun 17 22:47:08 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Fri, 18 Jun 2010 07:47:08 +0200 Subject: [Freeswitch-users] FS as Media Gateway Only In-Reply-To: References: <22AFCE13-56B2-44E0-861D-0D913D926DD8@gmail.com> <72EFF2C0-346D-44D4-8FB7-B3A0537D097C@jerris.com> Message-ID: <8A092D4F-E060-4811-8A57-B4DECCBA33A4@gmail.com> I don't see the issue. They will send their RTP to FS-RTP-x, and FS-RTP-x will autoadjust to this stream in order to learn the real IP:port of the endpoint. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 18/06/2010 ? 00:46, Code Ghar a ?crit : > I haven't had a chance to actually test this config in lab but I > will in the coming days. One issue I see is when we use registered > endpoints, such as ATAs, behind NAT: bypass_media fails in this case > because FS uses the private network IP instead of the public network > IP in SDP. In case of a large number of such registered endpoints, > we may have to make them talk with an FS-RTP server directly because > they don't use bypass_media while FS-SIP does. Can we use > bypass_media_after_reinvite for them or would it have the same > behavior of using private IP? Or we could use proxy_media; which is > not want we really want to do but could work in a pinch. Or is there > a better way to combine bypass_media with NAT'ed endpoints? > > > > On Mon, Jun 7, 2010 at 6:29 AM, David Ponzone > wrote: > Mike, > > You're right, it can be achieved with SIP now that I think a bit > more about it. > The idea was to allow adding multiple media gateways when required, > so the media gateways should not be facing the carriers as some of > them do SIP-filtering, but should only be advertised in the SDP. > > So SIP-only boxes (doing bypass-media) should face the carriers to > handle the trunking. > In the middle, we would then have the media gateways, doing SIP and > mostly RTP. > But I guess we dont want customers to register and to send calls to > a media gateway, so we need another set of SIP boxes on the other > side, doing bypass-media also. > > So it would like this: > > ------sip-----FS-RTP-1-----sip------ > FS-SIP-Internal-1 ------sip-----FS-RTP-2-----sip------FS-SIP- > External-1----sip-----Carriers > ------sip-----FS-RTP-3-----sip------ > FS-SIP-Internal-2 -------sip----FS-RTP-4-----sip------FS-SIP- > External-2-----sip----Carriers > -------sip----FS-RTP-5-----sip------ > > Thanks to bypass-media, the RTP streams would go from customer to FS- > RTP-x to Carriers, and reverse. > And I don't see any reason why the same set of FS-SIP boxes could > not be used for both internal and external borders. > > Is there something wrong in this ? > > Code, does it help ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 05/06/2010 ? 19:54, Michael Jerris a ?crit : > >> Why would it be an advantage to have your media proxies use another >> protocol? >> >> Mike >> >> On Jun 4, 2010, at 1:59 AM, David Ponzone wrote: >> >>> It doesn't solve the issue that all the media servers will do >>> signaling too, and will talk SIP with the carriers. >>> So the carriers will need to allow all the media servers . >>> >>> The only clean solution to avoid that, I think, is to have >>> signaling boxes allocating resources from media servers with >>> another protocol than SIP. >>> RTPproxy does that I think, but I am not sure how it works. >>> >>> David Ponzone >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/700a4735/attachment-0001.html From lakindia89 at gmail.com Thu Jun 17 22:50:47 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Fri, 18 Jun 2010 11:20:47 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: Dear Antony, Also in the leg_timeout wiki http://wiki.freeswitch.org/wiki/Variable_leg_timeout, it is stated as follows "If you are using group confirm then you can cancel the timeout by using the group_confirm_cancel_timeoutchannel variable." On Thu, Jun 17, 2010 at 8:22 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > no there is no way, besides making both timeouts longer. > you could file a feature request/bounty to ask for a feature to stop the > leg timer when you reach the confirm. > > > On Thu, Jun 17, 2010 at 4:23 AM, Nagalenoj H. wrote: > >> Anthony, >> But, then there is no use. Am I right? Usually, we'll use the >> group_confirm_cancel_timeout only when we need to override the leg_timeout. >> But it happens in reverse in this case., >> >> I've tried using the group_confirm_cancel_timeout along with call_timeout >> and things happening similar like setting leg_timout. >> >> Then, tried without setting leg_timeout and call_timeout explicitly. >> * In this case if the callee doesn't picks the call, it >> disconnects the leg in 30 secs. >> * If he answers the call and the script continues to execute, the >> leg is disconnected in 60 secs. >> >> What I need to do is, when the callee picks the call the leg_timeout >> should not be accounted more and the leg shouldn't be disconnected because >> of leg_timeout after that. >> >> Any other way of doing this?! >> >> >> >> On Tue, Jun 15, 2010 at 10:53 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> leg timeout beats the group confirm timeouts >>> >>> >>> On Tue, Jun 15, 2010 at 12:28 AM, Nagalenoj H. wrote: >>> >>>> Dear friends, >>>> I've tried using the group_confirm_cancel_timeout channel variable. >>>> I've written a testing script to get digits before bridging. But, it doesn't >>>> seem to be working. >>>> >>>> My understanding after reading wiki is, >>>> * When I dial [leg_timeout=10]user/1005, if he answers before >>>> timeout and in the process of giving digits, then the call shouldn't be >>>> disconnected after the leg_timeout secs (10 sec in the example). >>>> >>>> But, When I experiment it, the call is getting disconnected after 10 >>>> seconds and it doesn't bother whether the callee has answered the >>>> call(Started giving digits) or not answered at all. >>>> >>>> I've checked it with nc as follows, >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: set >>>> execute-app-arg: group_confirm_key=exec >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: set >>>> execute-app-arg: group_confirm_file=perl /root/confirm.pl >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: set >>>> execute-app-arg: group_confirm_cancel_timeout=1 >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: bridge >>>> execute-app-arg: [leg_timeout=10]user/1005 >>>> >>>> And here is the script, >>>> >>>> use freeswitch; >>>> our $session; >>>> my $digit; >>>> >>>> while(1) { >>>> # Wait till response timeout for the first digit. >>>> $digit = $session->getDigits(1, "", 10000); >>>> freeswitch::consoleLog ("info","Digit>>".$digit."<<"); >>>> >>>> if (! $session->ready() ) { >>>> freeswitch::consoleLog("info","Going to Exit\n"); >>>> last; >>>> } >>>> if (defined $digit and $digit ne "" ) { >>>> freeswitch::consoleLog("info","DTMF received: >>>> $digit\n"); >>>> if ($digit eq '#') { >>>> return; >>>> } >>>> } >>>> else { >>>> freeswitch::consoleLog("info","Timeout\n"); >>>> $session->hangup(); >>>> } >>>> } >>>> 1; >>>> >>>> If my understanding is right then, I believe there is something wrong >>>> with channel_variable. >>>> >>>> Kindly help me to resolve this. >>>> >>>> -- >>>> Regards, >>>> Nagalenoj H. >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Regards, >> Nagalenoj H. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/db53ed84/attachment.html From xengelpublicx at gmail.com Fri Jun 18 00:35:55 2010 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Fri, 18 Jun 2010 11:35:55 +0400 Subject: [Freeswitch-users] presence in linksys spa932 In-Reply-To: References: Message-ID: i'm change dialplan with call_limit: then the state spa932 phone for incoming calls stopped working. Always light green. On Thu, Jun 17, 2010 at 4:57 PM, Vladimir Elizarov wrote: > ?I found a bug in the presence of the linksys spa932 (configure this > article: http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932). > If during the > conversation (lamp on) to the subscriber, someone called, the lamp on > the panel goes into the state off. Who can verify whether he has > reproduced the same problem? > > Unit key: fnc=blf+sd+cp;sub=110@$PROXY > > freeswitch 1.0.6 (git 10 06 2010) > > Linksys spa962: > Software Version: ? ? ? 6.1.3(a) > Hardware Version: ? ? ? 1.0.3(917f) > > Linksys spa932 > Unit Enable: ? ?Yes ? ? Unit Online: ? ?Yes > Subscribe Expires: ? ? ?600 ? ? Subscribe Retry Interval: ? ? ? 6 > HW Version: ? ? 1.0.6 ? SW Version: ? ? 2.0.2 > > Configure image: > http://img192.imageshack.us/img192/2320/linksysspa932.png > > -- > Best regards, Vladimir Elizarov > -- Best regards, Vladimir Elizarov From tayeb.meftah at gmail.com Sat Jun 19 00:28:07 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sat, 19 Jun 2010 09:28:07 +0200 Subject: [Freeswitch-users] Build the mod_unimrcp module In-Reply-To: References: Message-ID: <4C1C7187.40409@gmail.com> hi, you need to customise the example extension to your need including the mrcp profile / tts voice name/... Le 18/06/2010 06:56, Thangappan.M a ?crit : > Dear all, > > I am in the process of developing IVR. So just planned to > convert my application to handle the TTS voice engine which is > supported by FreeSWITCH. > Got the mod_unimrcp module which is used to recognize the > speech and synthesize the text to voice. > For building the mod_unimrcp modules done the following steps. > Uncomment the mod_unimrcp line in the modules.conf file > in the FreeSWITCH source > Given make mod_unimrcp-install command. > In modules.conf.xml uncomment the modue="mod_unimrcp"/> > Configured the following dial plan > > > > > > > --> > > > > > While making the call to 4922 got the following error in the > FreeSWITCH console. > > > [INFO] mod_dialplan_xml.c:418 Processing thangappan->4922 in context default > [NOTICE] mod_dptools.c:717 Channel [sofia/internal/1012 at 192.168.1.222 ] has been answered > > [INFO] mod_unimrcp.c:1499 speech_handle: name = unimrcp, rate = 8000, speed = 0, samples = 160, voice = , engine = unimrcp, param = nuance5-mrcp1 > [INFO] mod_unimrcp.c:1502 voice = awb, rate = 8000 > [NOTICE] mrcp_client.c:549 Create MRCP Handle 0x8b02400 [nuance5-mrcp1] > > [INFO] mrcp_client_session.c:142 Create Channel 0x8b02400 > [INFO] mrcp_client_session.c:398 Receive App Request 0x8b02400 [2] > [NOTICE] rtsp_client.c:255 Create RTSP Handle 0x8b04408 > [INFO] mrcp_client.c:901 Add MRCP Handle 0x8b02400 > > [NOTICE] mrcp_client_session.c:718 Add Control Channel 0x8b02400 > [INFO] mrcp_client_session.c:420 Send Offer 0x8b02400 [c:0 a:1 v:0] > *[ERR] mod_unimrcp.c:965 (TTS-0) Timed out waiting for channel to be ready > > [ERR] switch_ivr_play_say.c:2104 Invalid TTS module!* > [NOTICE] switch_core_state_machine.c:185 sofia/internal/1012 at 192.168.1.222 has executed the last dialplan instruction, hanging up. > > [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1012 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING] > [NOTICE] switch_core_session.c:1179 Session 1 (sofia/internal/1012 at 192.168.1.222 ) Ended > > [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/1012 at 192.168.1.222 [CS_DESTROY] > [NOTICE] switch_channel.c:669 New Channel sofia/internal/1012 at 192.168.1.222 [49ed5afa-79f0-11df-b531-3553f3a65c3c] > > So need to find a solution for that. > > -- > Regards, > Thangappan.M > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100619/80e01d1c/attachment-0001.html From jerome.meuret at gmail.com Fri Jun 18 01:51:34 2010 From: jerome.meuret at gmail.com (=?ISO-8859-1?B?Suly9G1lIE0u?=) Date: Fri, 18 Jun 2010 10:51:34 +0200 Subject: [Freeswitch-users] Re : How to enable mod_skypopen without sound card Message-ID: On Windows, it exists Virtual Audio Cable to emulate a sound card, here is the website : http://software.muzychenko.net/eng/vac.html But it's not free... 2010/6/17 > 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. How to enable mod_skypopen without sound card (k xd) > 2. Aastra SCA Woes, Any Updates? (Wes Pearce) > 3. Re: How to enable mod_skypopen without sound card > (Giovanni Maruzzelli) > 4. Issue with an IVR-menu (Milena) > 5. Re: Aastra SCA Woes, Any Updates? (Brian West) > 6. Re: Aastra SCA Woes, Any Updates? (Chris Chen) > 7. Re: Aastra SCA Woes, Any Updates? (Brian West) > 8. Re: Issue with an IVR-menu (Milena) > 9. Re: Aastra SCA Woes, Any Updates? (Wes Pearce) > > > ---------- Message transf?r? ---------- > From: k xd > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 17 Jun 2010 16:40:47 +0800 > Subject: [Freeswitch-users] How to enable mod_skypopen without sound card > Hi, > > I setup freeswitch in a window server and then startup 2 skype clients, > however when I call registered skype user, I got this message "Contact can > only receive IMs". The server doesn't have sound card. So I guess that when > registered skype users try to transfer the incoming call, because of no > sound card it terminated the call directly. > > Does anyone know how to solve this issue? > > Thanks, > Will > > > ---------- Message transf?r? ---------- > From: Wes Pearce > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 17 Jun 2010 10:32:37 -0700 (PDT) > Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? > > Hello Users, > > I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in the > office here. I've been trying to get SCA working unsuccessfully. While I > was > digging around, I found this thread: > > http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html > > Which describes my exact problem. SCA works neatly on outbound calls, and > SCA works on incoming calls until the phone is answered... at which point > the call dissapears from the other lines. > > Brian West mentioned this has something to do with Aastras inconsistent > call-info headers. > > I've been speaking with a support representative from Aastra named Brian > Epps, who says their engineers are unaware of any SCA problems. > > Here are the appropriate SIP traces: > > incoming call: http://pastebin.com/QK9K3KL9 > outgoing call: http://pastebin.com/zzxbm9mq > > There are call info headers in both traces. It's just the call-info headers > in the incoming trace look incorrect. > > I'm wondering if there has been any movement on this, or if I've just > screwed something up somewhere. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp5191927p5191927.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > ---------- Message transf?r? ---------- > From: Giovanni Maruzzelli > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 17 Jun 2010 19:52:22 +0200 > Subject: Re: [Freeswitch-users] How to enable mod_skypopen without sound > card > On Thu, Jun 17, 2010 at 10:40 AM, k xd wrote: > > Hi, > > I setup freeswitch in a window server and then startup 2 skype clients, > > however when I call registered skype user, I got this message "Contact > can > > only receive IMs". The server doesn't have sound card. So I guess that > when > > registered skype users try to transfer the incoming call, because of no > > sound card it terminated the call directly. > > Does anyone know how to solve this issue? > > A windows server has no way (that I know) to fake a soundcard (as > opposed to the snd-dummy "fake" audio driver in Linux), and the Skype > clients check if there is a soundcard available, and refuse to do > audio if a soundcard is not available (is stupid, but is like that). > > So, your only option is to add a soundcard to the server, a cheap > dongle format USB soundcard would do as well (the soundcard is not > used for anything, but the Skype client check about its existance). > > -giovanni > > > > Thanks, > > Will > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > > > > ---------- Message transf?r? ---------- > From: Milena > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 17 Jun 2010 12:56:46 -0500 > Subject: [Freeswitch-users] Issue with an IVR-menu > Hello, > > Does anybody know why would freeswitch say ... switch_ivr_menu.c:851 Unable > to build xml menu ... mod_dptools.c:1264 Unable to create menu when trying > to call an IVR that was working fine before I updated this morning? ... and > how to fix it? ;) > > http://pastebin.freeswitch.org/13209 > > thank you > > > ---------- Message transf?r? ---------- > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 17 Jun 2010 13:12:43 -0500 > Subject: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? > The issue is the absence of the extra bits in the call-info header on every > packet involved in the call transaction... but its worse than that if you > want SCA you can only have ONE account on the phone and HOPE and pray it > behaves. I have tried to get Aastra to fix it but as far as I'm concerned > your best bet is to call Polycom and trade all those Aastra's for Polycom's > they have a trade up program and all you have to do is take a baseball bat > and make the Aastra's inoperable or not they seems to have the inoperable > part covered already right out of the box. > > /b > > On Jun 17, 2010, at 12:32 PM, Wes Pearce wrote: > > > > > Hello Users, > > > > I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in the > > office here. I've been trying to get SCA working unsuccessfully. While I > was > > digging around, I found this thread: > > > > > http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html > > > > Which describes my exact problem. SCA works neatly on outbound calls, and > > SCA works on incoming calls until the phone is answered... at which point > > the call dissapears from the other lines. > > > > Brian West mentioned this has something to do with Aastras inconsistent > > call-info headers. > > > > I've been speaking with a support representative from Aastra named Brian > > Epps, who says their engineers are unaware of any SCA problems. > > > > Here are the appropriate SIP traces: > > > > incoming call: http://pastebin.com/QK9K3KL9 > > outgoing call: http://pastebin.com/zzxbm9mq > > > > There are call info headers in both traces. It's just the call-info > headers > > in the incoming trace look incorrect. > > > > I'm wondering if there has been any movement on this, or if I've just > > screwed something up somewhere. > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp5191927p5191927.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > > ---------- Message transf?r? ---------- > From: Chris Chen > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 17 Jun 2010 14:31:34 -0400 > Subject: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? > I second that Polycom SCA is working perfectly reliably with multiple > accounts (if you have multiple lines such as Polycom IP 650). > Thanks, > Chris > > On Thu, Jun 17, 2010 at 2:12 PM, Brian West wrote: > >> The issue is the absence of the extra bits in the call-info header on >> every packet involved in the call transaction... but its worse than that if >> you want SCA you can only have ONE account on the phone and HOPE and pray it >> behaves. I have tried to get Aastra to fix it but as far as I'm concerned >> your best bet is to call Polycom and trade all those Aastra's for Polycom's >> they have a trade up program and all you have to do is take a baseball bat >> and make the Aastra's inoperable or not they seems to have the inoperable >> part covered already right out of the box. >> >> /b >> >> On Jun 17, 2010, at 12:32 PM, Wes Pearce wrote: >> >> > >> > Hello Users, >> > >> > I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in >> the >> > office here. I've been trying to get SCA working unsuccessfully. While I >> was >> > digging around, I found this thread: >> > >> > >> http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html >> > >> > Which describes my exact problem. SCA works neatly on outbound calls, >> and >> > SCA works on incoming calls until the phone is answered... at which >> point >> > the call dissapears from the other lines. >> > >> > Brian West mentioned this has something to do with Aastras inconsistent >> > call-info headers. >> > >> > I've been speaking with a support representative from Aastra named Brian >> > Epps, who says their engineers are unaware of any SCA problems. >> > >> > Here are the appropriate SIP traces: >> > >> > incoming call: http://pastebin.com/QK9K3KL9 >> > outgoing call: http://pastebin.com/zzxbm9mq >> > >> > There are call info headers in both traces. It's just the call-info >> headers >> > in the incoming trace look incorrect. >> > >> > I'm wondering if there has been any movement on this, or if I've just >> > screwed something up somewhere. >> > -- >> > View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp5191927p5191927.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > ---------- Message transf?r? ---------- > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 17 Jun 2010 13:40:14 -0500 > Subject: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? > The Linksys/CiscoSPA's do too :P And those interop between each other > correctly. > > /b > > On Jun 17, 2010, at 1:31 PM, Chris Chen wrote: > > > I second that Polycom SCA is working perfectly reliably with multiple > accounts (if you have multiple lines such as Polycom IP 650). > > Thanks, > > Chris > > > > On Thu, Jun 17, 2010 at 2:12 PM, Brian West > wrote: > > The issue is the absence of the extra bits in the call-info header on > every packet involved in the call transaction... but its worse than that if > you want SCA you can only have ONE account on the phone and HOPE and pray it > behaves. I have tried to get Aastra to fix it but as far as I'm concerned > your best bet is to call Polycom and trade all those Aastra's for Polycom's > they have a trade up program and all you have to do is take a baseball bat > and make the Aastra's inoperable or not they seems to have the inoperable > part covered already right out of the box. > > > > /b > > > > > > ---------- Message transf?r? ---------- > From: Milena > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 17 Jun 2010 14:16:26 -0500 > Subject: Re: [Freeswitch-users] Issue with an IVR-menu > The problem was the empty digits="" that was added when doing tests and the > menu hadn't been reloaded until i restarted freeswitch >_< > thank you anyways :) > > > > ---------- Message transf?r? ---------- > From: Wes Pearce > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 17 Jun 2010 12:43:16 -0700 (PDT) > Subject: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? > > Lame lame lame! > > Thanks though guys, you rock. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp5191927p5192528.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/5acd7d08/attachment-0001.html From abu.4000 at gmail.com Fri Jun 18 01:55:21 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Fri, 18 Jun 2010 14:25:21 +0530 Subject: [Freeswitch-users] Build the mod_unimrcp module In-Reply-To: <4C1C7187.40409@gmail.com> References: <4C1C7187.40409@gmail.com> Message-ID: Make sure you have installed any one of the following speech synthesizer , * UniMRCP Server * Speech Technology Center, VoiceNavigator * Nuance Speech Server 5.0/5.1 * Voxeo Prophecy 8.0 * Loquendo Suite 7.0 On Sat, Jun 19, 2010 at 12:58 PM, Meftah Tayeb wrote: > hi, > you need to customise the example extension to your need including the mrcp > profile / tts voice name/... > Le 18/06/2010 06:56, Thangappan.M a ?crit : > > Dear all, > > I am in the process of developing IVR. So just planned to convert > my application to handle the TTS voice engine which is supported by > FreeSWITCH. > Got the mod_unimrcp module which is used to recognize the speech > and synthesize the text to voice. > For building the mod_unimrcp modules done the following steps. > Uncomment the mod_unimrcp line in the modules.conf file in > the FreeSWITCH source > Given make mod_unimrcp-install command. > In modules.conf.xml uncomment the modue="mod_unimrcp"/> > Configured the following dial plan > name="unimrcp"> > > expression="^4922$"> > > > data="tts_engine=unimrcp:nuance5-mrcp1"/> > > --> > > > > > > > While making the call to 4922 got the following error in the > FreeSWITCH console. > > > > [INFO] mod_dialplan_xml.c:418 Processing thangappan->4922 in context default > [NOTICE] mod_dptools.c:717 Channel [sofia/internal/1012 at 192.168.1.222] has been answered > > [INFO] mod_unimrcp.c:1499 speech_handle: name = unimrcp, rate = 8000, speed = 0, samples = 160, voice = , engine = unimrcp, param = nuance5-mrcp1 > [INFO] mod_unimrcp.c:1502 voice = awb, rate = 8000 > [NOTICE] mrcp_client.c:549 Create MRCP Handle 0x8b02400 [nuance5-mrcp1] > > [INFO] mrcp_client_session.c:142 Create Channel 0x8b02400 > [INFO] mrcp_client_session.c:398 Receive App Request 0x8b02400 [2] > [NOTICE] rtsp_client.c:255 Create RTSP Handle 0x8b04408 > [INFO] mrcp_client.c:901 Add MRCP Handle 0x8b02400 > > [NOTICE] mrcp_client_session.c:718 Add Control Channel 0x8b02400 > [INFO] mrcp_client_session.c:420 Send Offer 0x8b02400 [c:0 a:1 v:0] > *[ERR] mod_unimrcp.c:965 (TTS-0) Timed out waiting for channel to be ready > > [ERR] switch_ivr_play_say.c:2104 Invalid TTS module!* > [NOTICE] switch_core_state_machine.c:185 sofia/internal/1012 at 192.168.1.222 has executed the last dialplan instruction, hanging up. > > [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1012 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING] > [NOTICE] switch_core_session.c:1179 Session 1 (sofia/internal/1012 at 192.168.1.222) Ended > > [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/1012 at 192.168.1.222 [CS_DESTROY] > [NOTICE] switch_channel.c:669 New Channel sofia/internal/1012 at 192.168.1.222 [49ed5afa-79f0-11df-b531-3553f3a65c3c] > > So need to find a solution for that. > > > -- > Regards, > Thangappan.M > > > _______________________________________________ > 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 > > > > -- > Meftah Tayeb > alg?rie t?l?com SPA > phone: +21321761805 > phone (INUM): +883510001289101 > mobile : +213660347746 > mobile (INUM: +883510001289110http://www.algerietelecom.dz > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 N.ABUBACKER SOFTWARE ENGINEER BK SYSTEMS (P) LTD CHENNAI-23 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/7e4584c0/attachment.html From xengelpublicx at gmail.com Fri Jun 18 02:47:32 2010 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Fri, 18 Jun 2010 13:47:32 +0400 Subject: [Freeswitch-users] presence in linksys spa932 In-Reply-To: References: Message-ID: debug presence 192.168.0.220 - spa962, 192.168.50.11 - fs 1.0.6, 192.168.0.176 - spa921, 10.8.6.6 - twinkle: spa921 ringing twinkle. spa921 status update, twinkle status not update ------------------------------------------------------------------------ send 937 bytes to udp/[192.168.0.220]:5060 at 09:29:28.504919: ------------------------------------------------------------------------ NOTIFY sip:100 at 192.168.0.220:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.11;rport;branch=z9hG4bK35tZKB43mmD7D Max-Forwards: 70 From: ;tag=Be64rve6FSU1D To: ;tag=beeaf51cf0d364d9 Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 CSeq: 132308348 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=599 Content-Type: application/dialog-info+xml Content-Length: 148 ------------------------------------------------------------------------ recv 285 bytes from udp/[192.168.0.220]:5060 at 09:29:28.516926: ------------------------------------------------------------------------ SIP/2.0 200 OK To: ;tag=beeaf51cf0d364d9 From: ;tag=Be64rve6FSU1D Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 CSeq: 132308348 NOTIFY Via: SIP/2.0/UDP 192.168.50.11;branch=z9hG4bK35tZKB43mmD7D Server: Linksys/SPA962-6.1.3(a) Content-Length: 0 ------------------------------------------------------------------------ freeswitch at 192.168.50.11@internal> recv 927 bytes from udp/[192.168.0.176]:5060 at 09:30:04.240224: ------------------------------------------------------------------------ INVITE sip:215 at tssec.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-cd8a0332 From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Nikolai Gabelok" Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 401 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 31175322 31175322 IN IP4 192.168.0.176 s=- c=IN IP4 192.168.0.176 t=0 0 m=audio 16472 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 321 bytes to udp/[192.168.0.176]:5060 at 09:30:04.240224: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-cd8a0332 From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2010-06-18 13:28:29.397241 [DEBUG] sofia.c:5949 IP 192.168.0.176 Rejected by acl "domains". Falling back to Digest auth. send 804 bytes to udp/[192.168.0.176]:5060 at 09:30:04.252231: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-cd8a0332 From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" ;tag=cQZXtQZ9c2HmS Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- 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 Proxy-Authenticate: Digest realm="tssec.ru", nonce="13679dc0-7abc-11df-97d3-6f7c82b8b4b4", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 407 bytes from udp/[192.168.0.176]:5060 at 09:30:04.276244: ------------------------------------------------------------------------ ACK sip:215 at tssec.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-cd8a0332 From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" ;tag=cQZXtQZ9c2HmS Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 101 ACK Max-Forwards: 70 Contact: "Nikolai Gabelok" User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ recv 1153 bytes from udp/[192.168.0.176]:5060 at 09:30:04.300257: ------------------------------------------------------------------------ INVITE sip:215 at tssec.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-5520e68e From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="229",realm="tssec.ru",nonce="13679dc0-7abc-11df-97d3-6f7c82b8b4b4",uri="sip:215 at tssec.ru",algorithm=MD5,response="78382843426af5a5fb18bc30de5953fb",qop=auth,nc=00000001,cnonce="4faa6b95" Contact: "Nikolai Gabelok" Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 401 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 31175322 31175322 IN IP4 192.168.0.176 s=- c=IN IP4 192.168.0.176 t=0 0 m=audio 16472 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 321 bytes to udp/[192.168.0.176]:5060 at 09:30:04.300257: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-5520e68e From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2010-06-18 13:28:29.461276 [DEBUG] sofia.c:5949 IP 192.168.0.176 Rejected by acl "domains". Falling back to Digest auth. 2010-06-18 13:28:29.465278 [NOTICE] switch_channel.c:772 New Channel sofia/internal/229 at tssec.ru [1371631e-7abc-11df-97d4-6f7c82b8b4b4] 2010-06-18 13:28:29.465278 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/229 at tssec.ru) Running State Change CS_NEW 2010-06-18 13:28:29.465278 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/229 at tssec.ru) State NEW 2010-06-18 13:28:29.469280 [DEBUG] sofia.c:4281 Channel sofia/internal/229 at tssec.ru entering state [received][100] 2010-06-18 13:28:29.469280 [DEBUG] sofia.c:4292 Remote SDP: v=0 o=- 31175322 31175322 IN IP4 192.168.0.176 s=- c=IN IP4 192.168.0.176 t=0 0 m=audio 16472 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2010-06-18 13:28:29.469280 [DEBUG] sofia_glue.c:3879 Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] 2010-06-18 13:28:29.469280 [DEBUG] sofia_glue.c:3879 Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] 2010-06-18 13:28:29.469280 [DEBUG] sofia_glue.c:2464 Set Codec sofia/internal/229 at tssec.ru PCMA/8000 20 ms 160 samples 2010-06-18 13:28:29.469280 [DEBUG] sofia_glue.c:3818 Set 2833 dtmf send/recv payload to 101 2010-06-18 13:28:29.469280 [DEBUG] sofia.c:4431 (sofia/internal/229 at tssec.ru) State Change CS_NEW -> CS_INIT 2010-06-18 13:28:29.469280 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/229 at tssec.ru [BREAK] 2010-06-18 13:28:29.469280 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/229 at tssec.ru) Running State Change CS_INIT 2010-06-18 13:28:29.469280 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/229 at tssec.ru) State INIT 2010-06-18 13:28:29.469280 [DEBUG] mod_sofia.c:83 sofia/internal/229 at tssec.ru SOFIA INIT 2010-06-18 13:28:29.469280 [DEBUG] mod_sofia.c:117 (sofia/internal/229 at tssec.ru) State Change CS_INIT -> CS_ROUTING 2010-06-18 13:28:29.469280 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/229 at tssec.ru [BREAK] 2010-06-18 13:28:29.469280 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/229 at tssec.ru) State INIT going to sleep 2010-06-18 13:28:29.469280 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/229 at tssec.ru) Running State Change CS_ROUTING 2010-06-18 13:28:29.469280 [DEBUG] switch_channel.c:1470 (sofia/internal/229 at tssec.ru) Callstate Change DOWN -> RINGING 2010-06-18 13:28:29.469280 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/229 at tssec.ru) State ROUTING 2010-06-18 13:28:29.469280 [DEBUG] switch_channel.c:1329 (sofia/internal/229 at tssec.ru) Callstate Change RINGING -> ACTIVE 2010-06-18 13:28:29.469280 [DEBUG] mod_sofia.c:140 sofia/internal/229 at tssec.ru SOFIA ROUTING 2010-06-18 13:28:29.469280 [DEBUG] switch_core_state_machine.c:77 sofia/internal/229 at tssec.ru Standard ROUTING 2010-06-18 13:28:29.469280 [INFO] mod_dialplan_xml.c:331 Processing Nikolai Gabelok->215 in context default Dialplan: sofia/internal/229 at tssec.ru parsing [default->unloop] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->time of day routing] continue=true Dialplan: sofia/internal/229 at tssec.ru Date/Time Match (PASS) [time of day routing] break=on-false Dialplan: sofia/internal/229 at tssec.ru Action set(status=open) Dialplan: sofia/internal/229 at tssec.ru Date/Time Match (FAIL) [time of day routing] break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->IVR] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [IVR] destination_number(215) =~ /^(ivr|5000)$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->delay_echo] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [delay_echo] destination_number(215) =~ /^echo123$|^123$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->delay_echo_next] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [delay_echo_next] destination_number(215) =~ /after_echo/ break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->global-intercept] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [global-intercept] destination_number(215) =~ /^\*\*$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->group-intercept] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [group-intercept] destination_number(215) =~ /^\*$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->extension-intercept] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [extension-intercept] destination_number(215) =~ /^\*(\d+)$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->sales] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [sales] destination_number(215) =~ /^sales$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->integrations] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [integrations] destination_number(215) =~ /^integrations$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->operators] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [operators] destination_number(215) =~ /^operators$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/229 at tssec.ru Regex (PASS) [Local_Extension] destination_number(215) =~ /^([1-2][0-2][0-9])$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru Action set(dialed_extension=215) Dialplan: sofia/internal/229 at tssec.ru Action export(dialed_extension=215) Dialplan: sofia/internal/229 at tssec.ru Action export(RECORD_STEREO=true) Dialplan: sofia/internal/229 at tssec.ru Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/229 at tssec.ru Action bind_meta_app(2 a s record_session::/var/lib/freeswitch/recordings/${caller_id_number}.${destination_number}.${strftime(%Y-%m-%d-%H-%M)}.wav) Dialplan: sofia/internal/229 at tssec.ru Action record_session(/var/lib/freeswitch/recordings/${caller_id_number}.${destination_number}.${strftime(%Y-%m-%d-%H-%M)}.wav) Dialplan: sofia/internal/229 at tssec.ru Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/229 at tssec.ru Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/229 at tssec.ru Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/229 at tssec.ru Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [Local_Extension] destination_number(215) =~ /^100$/ break=on-true Dialplan: sofia/internal/229 at tssec.ru Regex (FAIL) [Local_Extension] destination_number(215) =~ /^229$/ break=on-false Dialplan: sofia/internal/229 at tssec.ru ANTI-Action limit(tssec.ru ${destination_number} 1) Dialplan: sofia/internal/229 at tssec.ru ANTI-Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/229 at tssec.ru ANTI-Action set(ringback=%(800,3200,425,0)) Dialplan: sofia/internal/229 at tssec.ru ANTI-Action set(call_timeout=15) Dialplan: sofia/internal/229 at tssec.ru ANTI-Action set(hangup_after_bridge=true) Dialplan: sofia/internal/229 at tssec.ru ANTI-Action set(continue_on_fail=true) Dialplan: sofia/internal/229 at tssec.ru ANTI-Action bridge({ignore_early_media=true}${sofia_contact(internal/${dialed_extension}@tssec.ru)}) Dialplan: sofia/internal/229 at tssec.ru ANTI-Action transfer(not_answer-${dialed_extension} XML default) 2010-06-18 13:28:29.473282 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/229 at tssec.ru) State Change CS_ROUTING -> CS_EXECUTE 2010-06-18 13:28:29.473282 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/229 at tssec.ru [BREAK] 2010-06-18 13:28:29.473282 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/229 at tssec.ru) State ROUTING going to sleep 2010-06-18 13:28:29.473282 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/229 at tssec.ru) Running State Change CS_EXECUTE 2010-06-18 13:28:29.473282 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/229 at tssec.ru) State EXECUTE 2010-06-18 13:28:29.473282 [DEBUG] mod_sofia.c:233 sofia/internal/229 at tssec.ru SOFIA EXECUTE 2010-06-18 13:28:29.473282 [DEBUG] switch_core_state_machine.c:157 sofia/internal/229 at tssec.ru Standard EXECUTE EXECUTE sofia/internal/229 at tssec.ru set(status=open) 2010-06-18 13:28:29.473282 [DEBUG] mod_dptools.c:834 sofia/internal/229 at tssec.ru SET [status]=[open] EXECUTE sofia/internal/229 at tssec.ru set(dialed_extension=215) 2010-06-18 13:28:29.473282 [DEBUG] mod_dptools.c:834 sofia/internal/229 at tssec.ru SET [dialed_extension]=[215] EXECUTE sofia/internal/229 at tssec.ru export(dialed_extension=215) 2010-06-18 13:28:29.473282 [DEBUG] mod_dptools.c:918 EXPORT [dialed_extension]=[215] EXECUTE sofia/internal/229 at tssec.ru export(RECORD_STEREO=true) 2010-06-18 13:28:29.473282 [DEBUG] mod_dptools.c:918 EXPORT [RECORD_STEREO]=[true] EXECUTE sofia/internal/229 at tssec.ru bind_meta_app(1 b s execute_extension::dx XML features) 2010-06-18 13:28:29.473282 [INFO] switch_ivr_async.c:2429 Bound B-Leg: 1 execute_extension::dx XML features EXECUTE sofia/internal/229 at tssec.ru bind_meta_app(2 a s record_session::/var/lib/freeswitch/recordings/229.215.2010-06-18-13-28.wav) 2010-06-18 13:28:29.473282 [INFO] switch_ivr_async.c:2422 Bound A-Leg: 2 record_session::/var/lib/freeswitch/recordings/229.215.2010-06-18-13-28.wav 2010-06-18 13:28:29.473282 [DEBUG] switch_core_session.c:1763 Application record_session Requires media! pre_answering channel sofia/internal/229 at tssec.ru 2010-06-18 13:28:29.481286 [INFO] switch_core_session.c:1765 Sending early media 2010-06-18 13:28:29.481286 [DEBUG] sofia_glue.c:2704 AUDIO RTP [sofia/internal/229 at tssec.ru] 192.168.50.11 port 24616 -> 192.168.0.176 port 16472 codec: 8 ms: 20 2010-06-18 13:28:29.481286 [DEBUG] switch_rtp.c:1373 Starting timer [soft] 160 bytes per 20ms 2010-06-18 13:28:29.481286 [INFO] sofia_presence.c:663 IN START_PRESENCE_SQL (internal) 2010-06-18 13:28:29.481286 [ERR] sofia_presence.c:672 DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','tssec.ru',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='229' and (sub_to_host='tssec.ru' or presence_hosts like '%tssec.ru%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: Event-Name: [PRESENCE_IN] Core-UUID: [9341fc9e-7a4d-11df-92e6-6f7c82b8b4b4] FreeSWITCH-Hostname: [sip1.lan.tssec.ru] FreeSWITCH-IPv4: [192.168.50.11] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2010-06-18 13:28:29] Event-Date-GMT: [Fri, 18 Jun 2010 09:28:29 GMT] Event-Date-Timestamp: [1276853309469280] Event-Calling-File: [switch_channel.c] Event-Calling-Function: [switch_channel_presence] Event-Calling-Line-Number: [575] Channel-State: [CS_ROUTING] Channel-Call-State: [DOWN] Channel-State-Number: [2] Channel-Name: [sofia/internal/229 at tssec.ru] Unique-ID: [1371631e-7abc-11df-97d4-6f7c82b8b4b4] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-Presence-ID: [229 at tssec.ru] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMA] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMA] Channel-Write-Codec-Rate: [8000] Caller-Username: [229] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Nikolai Gabelok] Caller-Caller-ID-Number: [229] Caller-Network-Addr: [192.168.0.176] Caller-ANI: [229] Caller-Destination-Number: [215] Caller-Unique-ID: [1371631e-7abc-11df-97d4-6f7c82b8b4b4] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/229 at tssec.ru] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1276853309465278] Caller-Channel-Created-Time: [1276853309465278] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] proto: [src/switch_channel.c] login: [src/switch_channel.c] from: [229 at tssec.ru] rpid: [unknown] status: [CS_ROUTING] event_type: [presence] alt_event_type: [dialog] presence-call-direction: [inbound] event_count: [0] 2010-06-18 13:28:29.481286 [NOTICE] sofia_presence.c:1126 SEND PRESENCE To: 100 at tssec.ru From: 229 at tssec.ru Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 Profile: internal [internal] 2010-06-18 13:28:29.481286 [WARNING] sofia_presence.c:1350 send payload: confirmed 2010-06-18 13:28:29.481286 [INFO] sofia_presence.c:682 IN END_PRESENCE_SQL (internal) 2010-06-18 13:28:29.481286 [WARNING] sofia_presence.c:593 tssec.ru is an alias, skipping 2010-06-18 13:28:29.481286 [WARNING] sofia_presence.c:600 external is passive, skipping 2010-06-18 13:28:29.481286 [INFO] sofia_presence.c:663 IN START_PRESENCE_SQL (internal) 2010-06-18 13:28:29.481286 [ERR] sofia_presence.c:672 DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_EXECUTE','unknown','tssec.ru',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='229' and (sub_to_host='tssec.ru' or presence_hosts like '%tssec.ru%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: Event-Name: [PRESENCE_IN] Core-UUID: [9341fc9e-7a4d-11df-92e6-6f7c82b8b4b4] FreeSWITCH-Hostname: [sip1.lan.tssec.ru] FreeSWITCH-IPv4: [192.168.50.11] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2010-06-18 13:28:29] Event-Date-GMT: [Fri, 18 Jun 2010 09:28:29 GMT] Event-Date-Timestamp: [1276853309473282] Event-Calling-File: [switch_channel.c] Event-Calling-Function: [switch_channel_presence] Event-Calling-Line-Number: [575] Channel-State: [CS_EXECUTE] Channel-Call-State: [ACTIVE] Channel-State-Number: [4] Channel-Name: [sofia/internal/229 at tssec.ru] Unique-ID: [1371631e-7abc-11df-97d4-6f7c82b8b4b4] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-Presence-ID: [229 at tssec.ru] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMA] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMA] Channel-Write-Codec-Rate: [8000] Caller-Username: [229] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Nikolai Gabelok] Caller-Caller-ID-Number: [229] Caller-Network-Addr: [192.168.0.176] Caller-ANI: [229] Caller-Destination-Number: [215] Caller-Unique-ID: [1371631e-7abc-11df-97d4-6f7c82b8b4b4] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/229 at tssec.ru] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1276853309465278] Caller-Channel-Created-Time: [1276853309465278] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] proto: [src/switch_channel.c] login: [src/switch_channel.c] from: [229 at tssec.ru] rpid: [unknown] status: [CS_EXECUTE] event_type: [presence] alt_event_type: [dialog] presence-call-direction: [inbound] event_count: [1] send 1043 bytes to udp/[192.168.0.220]:5060 at 09:30:04.324270: ------------------------------------------------------------------------ NOTIFY sip:100 at 192.168.0.220:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.11;rport;branch=z9hG4bK4emrN6m7HX3SS Max-Forwards: 70 From: ;tag=Be64rve6FSU1D To: ;tag=beeaf51cf0d364d9 Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 CSeq: 132308349 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1164 Content-Type: application/dialog-info+xml Content-Length: 253 confirmed ------------------------------------------------------------------------ 2010-06-18 13:28:29.481286 [NOTICE] sofia_presence.c:1126 SEND PRESENCE To: 100 at tssec.ru From: 229 at tssec.ru Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 Profile: internal [internal] 2010-06-18 13:28:29.481286 [DEBUG] sofia_glue.c:2914 Set 2833 dtmf send payload to 101 2010-06-18 13:28:29.481286 [DEBUG] sofia_glue.c:2919 Set 2833 dtmf receive payload to 101 2010-06-18 13:28:29.481286 [DEBUG] mod_sofia.c:2101 Ring SDP: v=0 o=FreeSWITCH 1276828693 1276828694 IN IP4 192.168.50.11 s=FreeSWITCH c=IN IP4 192.168.50.11 t=0 0 m=audio 24616 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-06-18 13:28:29.481286 [NOTICE] mod_sofia.c:2104 Pre-Answer sofia/internal/229 at tssec.ru! 2010-06-18 13:28:29.481286 [INFO] sofia_presence.c:682 IN END_PRESENCE_SQL (internal) 2010-06-18 13:28:29.481286 [WARNING] sofia_presence.c:593 tssec.ru is an alias, skipping 2010-06-18 13:28:29.481286 [WARNING] sofia_presence.c:600 external is passive, skipping 2010-06-18 13:28:29.481286 [DEBUG] switch_channel.c:2343 (sofia/internal/229 at tssec.ru) Callstate Change ACTIVE -> EARLY 2010-06-18 13:28:29.481286 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/229 at tssec.ru [BREAK] EXECUTE sofia/internal/229 at tssec.ru record_session(/var/lib/freeswitch/recordings/229.215.2010-06-18-13-28.wav) 2010-06-18 13:28:29.481286 [DEBUG] switch_core_media_bug.c:365 Attaching BUG to sofia/internal/229 at tssec.ru EXECUTE sofia/internal/229 at tssec.ru hash(insert/tssec.ru-call_return/215/229) send 1115 bytes to udp/[192.168.0.176]:5060 at 09:30:04.324270: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-5520e68e From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" ;tag=D0rpvjgDaB86m Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 102 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- 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-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "215" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1276828693 1276828694 IN IP4 192.168.50.11 s=FreeSWITCH c=IN IP4 192.168.50.11 t=0 0 m=audio 24616 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2010-06-18 13:28:29.481286 [DEBUG] sofia.c:4276 Channel sofia/internal/229 at tssec.ru skipping state [early][183] EXECUTE sofia/internal/229 at tssec.ru hash(insert/tssec.ru-last_dial_ext/215/1371631e-7abc-11df-97d4-6f7c82b8b4b4) EXECUTE sofia/internal/229 at tssec.ru set(called_party_callgroup=1) 2010-06-18 13:28:29.481286 [DEBUG] mod_dptools.c:834 sofia/internal/229 at tssec.ru SET [called_party_callgroup]=[1] EXECUTE sofia/internal/229 at tssec.ru hash(insert/tssec.ru-last_dial/1/1371631e-7abc-11df-97d4-6f7c82b8b4b4) EXECUTE sofia/internal/229 at tssec.ru limit(tssec.ru 215 1) 2010-06-18 13:28:29.481286 [INFO] mod_limit.c:729 Usage for tssec.ru_215 is now 1/1 EXECUTE sofia/internal/229 at tssec.ru set(transfer_ringback=local_stream://moh) 2010-06-18 13:28:29.493293 [DEBUG] mod_dptools.c:834 sofia/internal/229 at tssec.ru SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/229 at tssec.ru set(ringback=%(800,3200,425,0)) 2010-06-18 13:28:29.493293 [DEBUG] mod_dptools.c:834 sofia/internal/229 at tssec.ru SET [ringback]=[%(800,3200,425,0)] EXECUTE sofia/internal/229 at tssec.ru set(call_timeout=15) 2010-06-18 13:28:29.493293 [DEBUG] mod_dptools.c:834 sofia/internal/229 at tssec.ru SET [call_timeout]=[15] EXECUTE sofia/internal/229 at tssec.ru set(hangup_after_bridge=true) 2010-06-18 13:28:29.493293 [DEBUG] mod_dptools.c:834 sofia/internal/229 at tssec.ru SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/229 at tssec.ru set(continue_on_fail=true) 2010-06-18 13:28:29.493293 [DEBUG] mod_dptools.c:834 sofia/internal/229 at tssec.ru SET [continue_on_fail]=[true] EXECUTE sofia/internal/229 at tssec.ru bridge({ignore_early_media=true}sofia/internal/sip:215 at 10.8.6.6:5080,sofia/internal/sip:215 at 192.168.1.118:5060) 2010-06-18 13:28:29.493293 [DEBUG] switch_ivr_originate.c:1944 variable string 0 = [ignore_early_media=true] 2010-06-18 13:28:29.493293 [NOTICE] switch_channel.c:772 New Channel sofia/internal/sip:215 at 10.8.6.6:5080 [1375a974-7abc-11df-97d5-6f7c82b8b4b4] 2010-06-18 13:28:29.493293 [DEBUG] mod_sofia.c:3817 (sofia/internal/sip:215 at 10.8.6.6:5080) State Change CS_NEW -> CS_INIT 2010-06-18 13:28:29.493293 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:29.493293 [NOTICE] switch_channel.c:772 New Channel sofia/internal/sip:215 at 192.168.1.118:5060 [1375a975-7abc-11df-97d6-6f7c82b8b4b4] 2010-06-18 13:28:29.493293 [DEBUG] mod_sofia.c:3817 (sofia/internal/sip:215 at 192.168.1.118:5060) State Change CS_NEW -> CS_INIT 2010-06-18 13:28:29.493293 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 192.168.1.118:5060 [BREAK] 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 192.168.1.118:5060) Running State Change CS_INIT 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 10.8.6.6:5080) Running State Change CS_INIT 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:215 at 10.8.6.6:5080) State INIT 2010-06-18 13:28:29.493293 [DEBUG] mod_sofia.c:83 sofia/internal/sip:215 at 10.8.6.6:5080 SOFIA INIT 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:215 at 192.168.1.118:5060) State INIT 2010-06-18 13:28:29.493293 [DEBUG] mod_sofia.c:83 sofia/internal/sip:215 at 192.168.1.118:5060 SOFIA INIT send 1185 bytes to udp/[10.8.6.6]:5080 at 09:30:04.332274: ------------------------------------------------------------------------ INVITE sip:215 at 10.8.6.6:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.11;rport;branch=z9hG4bK5QDHQ15aF6ScN Max-Forwards: 69 From: "Nikolai Gabelok" ;tag=e9HFyD1g7KySg To: Call-ID: ead3e978-f55e-122d-a480-00163efcbed2 CSeq: 132308366 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- 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-Type: application/sdp Content-Disposition: session Content-Length: 293 X-FS-Support: update_display Remote-Party-ID: "Nikolai Gabelok" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1276826303 1276826304 IN IP4 192.168.50.11 s=FreeSWITCH c=IN IP4 192.168.50.11 t=0 0 m=audio 27006 RTP/AVP 8 0 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2010-06-18 13:28:29.493293 [DEBUG] mod_sofia.c:117 (sofia/internal/sip:215 at 10.8.6.6:5080) State Change CS_INIT -> CS_ROUTING 2010-06-18 13:28:29.493293 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:215 at 10.8.6.6:5080) State INIT going to sleep 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 10.8.6.6:5080) Running State Change CS_ROUTING 2010-06-18 13:28:29.493293 [DEBUG] switch_channel.c:1470 (sofia/internal/sip:215 at 10.8.6.6:5080) Callstate Change DOWN -> RINGING 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:215 at 10.8.6.6:5080) State ROUTING 2010-06-18 13:28:29.493293 [DEBUG] switch_channel.c:1329 (sofia/internal/sip:215 at 10.8.6.6:5080) Callstate Change RINGING -> ACTIVE 2010-06-18 13:28:29.493293 [DEBUG] mod_sofia.c:140 sofia/internal/sip:215 at 10.8.6.6:5080 SOFIA ROUTING 2010-06-18 13:28:29.493293 [DEBUG] switch_ivr_originate.c:64 (sofia/internal/sip:215 at 10.8.6.6:5080) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-06-18 13:28:29.493293 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:215 at 10.8.6.6:5080) State ROUTING going to sleep 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 10.8.6.6:5080) Running State Change CS_CONSUME_MEDIA 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:215 at 10.8.6.6:5080) State CONSUME_MEDIA 2010-06-18 13:28:29.493293 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:215 at 10.8.6.6:5080) State CONSUME_MEDIA going to sleep recv 285 bytes from udp/[192.168.0.220]:5060 at 09:30:04.336276: ------------------------------------------------------------------------ SIP/2.0 200 OK To: ;tag=beeaf51cf0d364d9 From: ;tag=Be64rve6FSU1D Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 CSeq: 132308349 NOTIFY Via: SIP/2.0/UDP 192.168.50.11;branch=z9hG4bK4emrN6m7HX3SS Server: Linksys/SPA962-6.1.3(a) Content-Length: 0 ------------------------------------------------------------------------ 2010-06-18 13:28:29.493293 [DEBUG] sofia.c:4281 Channel sofia/internal/sip:215 at 10.8.6.6:5080 entering state [calling][0] 2010-06-18 13:28:29.497295 [DEBUG] mod_sofia.c:117 (sofia/internal/sip:215 at 192.168.1.118:5060) State Change CS_INIT -> CS_ROUTING 2010-06-18 13:28:29.497295 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 192.168.1.118:5060 [BREAK] 2010-06-18 13:28:29.497295 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:215 at 192.168.1.118:5060) State INIT going to sleep 2010-06-18 13:28:29.497295 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 192.168.1.118:5060) Running State Change CS_ROUTING 2010-06-18 13:28:29.497295 [DEBUG] switch_channel.c:1470 (sofia/internal/sip:215 at 192.168.1.118:5060) Callstate Change DOWN -> RINGING send 1195 bytes to udp/[192.168.1.118]:5060 at 09:30:04.336276: ------------------------------------------------------------------------ INVITE sip:215 at 192.168.1.118:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.11;rport;branch=z9hG4bK6069rvpecFgZg Max-Forwards: 69 From: "Nikolai Gabelok" ;tag=FjB8Z8Hm4vmcc To: Call-ID: ead485ce-f55e-122d-a480-00163efcbed2 CSeq: 132308366 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- 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-Type: application/sdp Content-Disposition: session Content-Length: 293 X-FS-Support: update_display Remote-Party-ID: "Nikolai Gabelok" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1276821601 1276821602 IN IP4 192.168.50.11 s=FreeSWITCH c=IN IP4 192.168.50.11 t=0 0 m=audio 31708 RTP/AVP 8 0 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2010-06-18 13:28:29.497295 [DEBUG] sofia.c:4281 Channel sofia/internal/sip:215 at 192.168.1.118:5060 entering state [calling][0] 2010-06-18 13:28:29.501297 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:215 at 192.168.1.118:5060) State ROUTING 2010-06-18 13:28:29.501297 [DEBUG] switch_channel.c:1329 (sofia/internal/sip:215 at 192.168.1.118:5060) Callstate Change RINGING -> ACTIVE 2010-06-18 13:28:29.501297 [DEBUG] mod_sofia.c:140 sofia/internal/sip:215 at 192.168.1.118:5060 SOFIA ROUTING 2010-06-18 13:28:29.501297 [DEBUG] switch_ivr_originate.c:64 (sofia/internal/sip:215 at 192.168.1.118:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-06-18 13:28:29.501297 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 192.168.1.118:5060 [BREAK] 2010-06-18 13:28:29.501297 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:215 at 192.168.1.118:5060) State ROUTING going to sleep 2010-06-18 13:28:29.501297 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 192.168.1.118:5060) Running State Change CS_CONSUME_MEDIA 2010-06-18 13:28:29.501297 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:215 at 192.168.1.118:5060) State CONSUME_MEDIA 2010-06-18 13:28:29.501297 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:215 at 192.168.1.118:5060) State CONSUME_MEDIA going to sleep recv 392 bytes from udp/[192.168.1.118]:5060 at 09:30:04.348283: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.11;rport=5060;branch=z9hG4bK6069rvpecFgZg From: "Nikolai Gabelok" ;tag=FjB8Z8Hm4vmcc To: Call-ID: ead485ce-f55e-122d-a480-00163efcbed2 CSeq: 132308366 INVITE Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE User-Agent: SIPPER for PhonerLite Content-Length: 0 ------------------------------------------------------------------------ recv 469 bytes from udp/[192.168.1.118]:5060 at 09:30:04.352285: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.50.11;rport=5060;branch=z9hG4bK6069rvpecFgZg From: "Nikolai Gabelok" ;tag=FjB8Z8Hm4vmcc To: ;tag=800c08bf2979df11bda80013d4bfa980 Call-ID: ead485ce-f55e-122d-a480-00163efcbed2 CSeq: 132308366 INVITE Contact: Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE User-Agent: SIPPER for PhonerLite Content-Length: 0 ------------------------------------------------------------------------ 2010-06-18 13:28:29.509302 [INFO] sofia.c:662 sofia/internal/sip:215 at 192.168.1.118:5060 Update Callee ID to "215" <215> 2010-06-18 13:28:29.521308 [DEBUG] sofia.c:4281 Channel sofia/internal/sip:215 at 192.168.1.118:5060 entering state [proceeding][180] 2010-06-18 13:28:29.521308 [NOTICE] sofia.c:4351 Ring-Ready sofia/internal/sip:215 at 192.168.1.118:5060! 2010-06-18 13:28:29.541319 [DEBUG] switch_rtp.c:2476 Correct ip/port confirmed. 2010-06-18 13:28:29.561329 [DEBUG] switch_ivr_originate.c:1125 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2010-06-18 13:28:29.561329 [DEBUG] switch_core_codec.c:122 sofia/internal/229 at tssec.ru Push codec L16:10 2010-06-18 13:28:29.561329 [DEBUG] switch_ivr_originate.c:1190 Play Ringback Tone [%(800,3200,425,0)] recv 325 bytes from udp/[10.8.6.6]:5080 at 09:30:04.440332: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.11;received=192.168.50.11;rport=5060;branch=z9hG4bK5QDHQ15aF6ScN To: From: "Nikolai Gabelok" ;tag=e9HFyD1g7KySg Call-ID: ead3e978-f55e-122d-a480-00163efcbed2 CSeq: 132308366 INVITE Server: Twinkle/1.4.2 Content-Length: 0 ------------------------------------------------------------------------ recv 370 bytes from udp/[10.8.6.6]:5080 at 09:30:04.440332: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.50.11;received=192.168.50.11;rport=5060;branch=z9hG4bK5QDHQ15aF6ScN To: ;tag=fyiai From: "Nikolai Gabelok" ;tag=e9HFyD1g7KySg Call-ID: ead3e978-f55e-122d-a480-00163efcbed2 CSeq: 132308366 INVITE Contact: Server: Twinkle/1.4.2 Content-Length: 0 ------------------------------------------------------------------------ 2010-06-18 13:28:29.601351 [INFO] sofia.c:662 sofia/internal/sip:215 at 10.8.6.6:5080 Update Callee ID to "215" <215> 2010-06-18 13:28:29.605353 [DEBUG] sofia.c:4281 Channel sofia/internal/sip:215 at 10.8.6.6:5080 entering state [proceeding][180] 2010-06-18 13:28:29.605353 [NOTICE] sofia.c:4351 Ring-Ready sofia/internal/sip:215 at 10.8.6.6:5080! recv 709 bytes from udp/[10.8.6.6]:5080 at 09:30:06.501439: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.11;received=192.168.50.11;rport=5060;branch=z9hG4bK5QDHQ15aF6ScN To: ;tag=fyiai From: "Nikolai Gabelok" ;tag=e9HFyD1g7KySg Call-ID: ead3e978-f55e-122d-a480-00163efcbed2 CSeq: 132308366 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Server: Twinkle/1.4.2 Supported: replaces,norefersub Content-Length: 199 v=0 o=twinkle 1275731453 434782481 IN IP4 10.8.6.6 s=- c=IN IP4 10.8.6.6 t=0 0 m=audio 16535 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 ------------------------------------------------------------------------ 2010-06-18 13:28:31.670463 [DEBUG] sofia.c:4281 Channel sofia/internal/sip:215 at 10.8.6.6:5080 entering state [completing][200] 2010-06-18 13:28:31.670463 [DEBUG] sofia.c:4292 Remote SDP: v=0 o=twinkle 1275731453 434782481 IN IP4 10.8.6.6 s=- c=IN IP4 10.8.6.6 t=0 0 m=audio 16535 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 send 359 bytes to udp/[10.8.6.6]:5080 at 09:30:06.509444: ------------------------------------------------------------------------ ACK sip:215 at 10.8.6.6:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.11;rport;branch=z9hG4bK79Z2tQ7H9Q6Hc Max-Forwards: 70 From: "Nikolai Gabelok" ;tag=e9HFyD1g7KySg To: ;tag=fyiai Call-ID: ead3e978-f55e-122d-a480-00163efcbed2 CSeq: 132308366 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2010-06-18 13:28:31.670463 [DEBUG] sofia.c:4281 Channel sofia/internal/sip:215 at 10.8.6.6:5080 entering state [ready][200] 2010-06-18 13:28:31.670463 [DEBUG] sofia_glue.c:3879 Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] 2010-06-18 13:28:31.670463 [DEBUG] sofia_glue.c:2464 Set Codec sofia/internal/sip:215 at 10.8.6.6:5080 PCMA/8000 20 ms 160 samples 2010-06-18 13:28:31.670463 [DEBUG] sofia_glue.c:3812 Set 2833 dtmf send payload to 101 2010-06-18 13:28:31.670463 [DEBUG] sofia_glue.c:2704 AUDIO RTP [sofia/internal/sip:215 at 10.8.6.6:5080] 192.168.50.11 port 27006 -> 10.8.6.6 port 16535 codec: 8 ms: 20 2010-06-18 13:28:31.670463 [DEBUG] switch_rtp.c:1373 Starting timer [soft] 160 bytes per 20ms 2010-06-18 13:28:31.670463 [DEBUG] sofia_glue.c:2914 Set 2833 dtmf send payload to 101 2010-06-18 13:28:31.670463 [DEBUG] sofia_glue.c:2919 Set 2833 dtmf receive payload to 101 2010-06-18 13:28:31.670463 [DEBUG] switch_channel.c:2497 Send signal sofia/internal/229 at tssec.ru [BREAK] 2010-06-18 13:28:31.670463 [NOTICE] sofia.c:4831 Channel [sofia/internal/sip:215 at 10.8.6.6:5080] has been answered 2010-06-18 13:28:31.682469 [DEBUG] switch_core_codec.c:146 sofia/internal/229 at tssec.ru Restore previous codec PCMA:8. 2010-06-18 13:28:31.682469 [DEBUG] switch_channel.c:2257 (sofia/internal/sip:215 at 192.168.1.118:5060) Callstate Change ACTIVE -> HANGUP 2010-06-18 13:28:31.682469 [NOTICE] switch_ivr_originate.c:3192 Hangup sofia/internal/sip:215 at 192.168.1.118:5060 [CS_CONSUME_MEDIA] [LOSE_RACE] 2010-06-18 13:28:31.682469 [DEBUG] switch_channel.c:2273 Send signal sofia/internal/sip:215 at 192.168.1.118:5060 [KILL] 2010-06-18 13:28:31.682469 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 192.168.1.118:5060 [BREAK] 2010-06-18 13:28:31.682469 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 192.168.1.118:5060) Running State Change CS_HANGUP 2010-06-18 13:28:31.682469 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/sip:215 at 192.168.1.118:5060) State HANGUP 2010-06-18 13:28:31.682469 [DEBUG] mod_sofia.c:441 Channel sofia/internal/sip:215 at 192.168.1.118:5060 hanging up, cause: LOSE_RACE 2010-06-18 13:28:31.690473 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/229 at tssec.ru: v=0 o=FreeSWITCH 1276828693 1276828695 IN IP4 192.168.50.11 s=FreeSWITCH c=IN IP4 192.168.50.11 t=0 0 m=audio 24616 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-06-18 13:28:31.690473 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/229 at tssec.ru [BREAK] 2010-06-18 13:28:31.690473 [DEBUG] switch_channel.c:2485 (sofia/internal/229 at tssec.ru) Callstate Change EARLY -> ACTIVE 2010-06-18 13:28:31.690473 [NOTICE] switch_ivr_originate.c:3216 Channel [sofia/internal/229 at tssec.ru] has been answered 2010-06-18 13:28:31.690473 [DEBUG] switch_ivr_originate.c:3261 Originate Resulted in Success: [sofia/internal/sip:215 at 10.8.6.6:5080] 2010-06-18 13:28:31.690473 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:31.690473 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/229 at tssec.ru [BREAK] 2010-06-18 13:28:31.690473 [DEBUG] switch_ivr_bridge.c:1182 (sofia/internal/sip:215 at 10.8.6.6:5080) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA send 1076 bytes to udp/[192.168.0.176]:5060 at 09:30:06.529454: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-5520e68e From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" ;tag=D0rpvjgDaB86m Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 102 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- 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-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "215" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1276828693 1276828694 IN IP4 192.168.50.11 s=FreeSWITCH c=IN IP4 192.168.50.11 t=0 0 m=audio 24616 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2010-06-18 13:28:31.690473 [DEBUG] sofia.c:4281 Channel sofia/internal/229 at tssec.ru entering state [completed][200] 2010-06-18 13:28:31.690473 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:31.690473 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 10.8.6.6:5080) Running State Change CS_EXCHANGE_MEDIA 2010-06-18 13:28:31.690473 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:215 at 10.8.6.6:5080) State EXCHANGE_MEDIA 2010-06-18 13:28:31.690473 [DEBUG] mod_sofia.c:534 SOFIA EXCHANGE_MEDIA 2010-06-18 13:28:31.702480 [DEBUG] mod_sofia.c:494 Sending CANCEL to sofia/internal/sip:215 at 192.168.1.118:5060 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:215 at 192.168.1.118:5060 Standard HANGUP, cause: LOSE_RACE 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/sip:215 at 192.168.1.118:5060) State HANGUP going to sleep 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/sip:215 at 192.168.1.118:5060) State Change CS_HANGUP -> CS_REPORTING 2010-06-18 13:28:31.702480 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 192.168.1.118:5060 [BREAK] 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 192.168.1.118:5060) Running State Change CS_REPORTING 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/sip:215 at 192.168.1.118:5060) State REPORTING 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:215 at 192.168.1.118:5060 Standard REPORTING, cause: LOSE_RACE 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/sip:215 at 192.168.1.118:5060) State REPORTING going to sleep 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/sip:215 at 192.168.1.118:5060) State Change CS_REPORTING -> CS_DESTROY 2010-06-18 13:28:31.702480 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 192.168.1.118:5060 [BREAK] 2010-06-18 13:28:31.702480 [DEBUG] switch_core_session.c:1171 Session 116 (sofia/internal/sip:215 at 192.168.1.118:5060) Locked, Waiting on external entities 2010-06-18 13:28:31.702480 [NOTICE] switch_core_session.c:1189 Session 116 (sofia/internal/sip:215 at 192.168.1.118:5060) Ended 2010-06-18 13:28:31.702480 [NOTICE] switch_core_session.c:1191 Close Channel sofia/internal/sip:215 at 192.168.1.118:5060 [CS_DESTROY] 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/sip:215 at 192.168.1.118:5060) Callstate Change HANGUP -> DOWN 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/sip:215 at 192.168.1.118:5060) Running State Change CS_DESTROY 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:215 at 192.168.1.118:5060) State DESTROY 2010-06-18 13:28:31.702480 [DEBUG] mod_sofia.c:350 sofia/internal/sip:215 at 192.168.1.118:5060 SOFIA DESTROY 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:215 at 192.168.1.118:5060 Standard DESTROY 2010-06-18 13:28:31.702480 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:215 at 192.168.1.118:5060) State DESTROY going to sleep send 375 bytes to udp/[192.168.1.118]:5060 at 09:30:06.541461: ------------------------------------------------------------------------ CANCEL sip:215 at 192.168.1.118:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.11;rport;branch=z9hG4bK6069rvpecFgZg Max-Forwards: 69 From: "Nikolai Gabelok" ;tag=FjB8Z8Hm4vmcc To: Call-ID: ead485ce-f55e-122d-a480-00163efcbed2 CSeq: 132308366 CANCEL Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 ------------------------------------------------------------------------ recv 472 bytes from udp/[192.168.1.118]:5060 at 09:30:06.549465: ------------------------------------------------------------------------ SIP/2.0 200 cancelling Via: SIP/2.0/UDP 192.168.50.11;rport=5060;branch=z9hG4bK6069rvpecFgZg From: "Nikolai Gabelok" ;tag=FjB8Z8Hm4vmcc To: ;tag=800c08bf2979df11bda80013d4bfa980 Call-ID: ead485ce-f55e-122d-a480-00163efcbed2 CSeq: 132308366 CANCEL Contact: Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE User-Agent: SIPPER for PhonerLite Content-Length: 0 ------------------------------------------------------------------------ recv 479 bytes from udp/[192.168.1.118]:5060 at 09:30:06.549465: ------------------------------------------------------------------------ SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.50.11;rport=5060;branch=z9hG4bK6069rvpecFgZg From: "Nikolai Gabelok" ;tag=FjB8Z8Hm4vmcc To: ;tag=800c08bf2979df11bda80013d4bfa980 Call-ID: ead485ce-f55e-122d-a480-00163efcbed2 CSeq: 132308366 INVITE Contact: Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE User-Agent: SIPPER for PhonerLite Content-Length: 0 ------------------------------------------------------------------------ send 351 bytes to udp/[192.168.1.118]:5060 at 09:30:06.549465: ------------------------------------------------------------------------ ACK sip:215 at 192.168.1.118:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.11;rport;branch=z9hG4bK6069rvpecFgZg Max-Forwards: 69 From: "Nikolai Gabelok" ;tag=FjB8Z8Hm4vmcc To: ;tag=800c08bf2979df11bda80013d4bfa980 Call-ID: ead485ce-f55e-122d-a480-00163efcbed2 CSeq: 132308366 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 657 bytes from udp/[192.168.0.176]:5060 at 09:30:06.561472: ------------------------------------------------------------------------ ACK sip:215 at 192.168.50.11:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-f67380cd From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" ;tag=D0rpvjgDaB86m Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="229",realm="tssec.ru",nonce="13679dc0-7abc-11df-97d3-6f7c82b8b4b4",uri="sip:215 at tssec.ru",algorithm=MD5,response="78382843426af5a5fb18bc30de5953fb",qop=auth,nc=00000001,cnonce="4faa6b95" Contact: "Nikolai Gabelok" User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ 2010-06-18 13:28:31.722491 [DEBUG] sofia.c:4281 Channel sofia/internal/229 at tssec.ru entering state [ready][200] 2010-06-18 13:28:31.742501 [DEBUG] switch_core_session.c:704 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:31.742501 [DEBUG] switch_core_session.c:704 Send signal sofia/internal/229 at tssec.ru [BREAK] 2010-06-18 13:28:31.782523 [INFO] sofia_presence.c:663 IN START_PRESENCE_SQL (internal) 2010-06-18 13:28:31.782523 [ERR] sofia_presence.c:672 DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'answered','unknown','tssec.ru',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='229' and (sub_to_host='tssec.ru' or presence_hosts like '%tssec.ru%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: Event-Name: [PRESENCE_IN] Core-UUID: [9341fc9e-7a4d-11df-92e6-6f7c82b8b4b4] FreeSWITCH-Hostname: [sip1.lan.tssec.ru] FreeSWITCH-IPv4: [192.168.50.11] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2010-06-18 13:28:31] Event-Date-GMT: [Fri, 18 Jun 2010 09:28:31 GMT] Event-Date-Timestamp: [1276853311690473] Event-Calling-File: [switch_channel.c] Event-Calling-Function: [switch_channel_presence] Event-Calling-Line-Number: [575] Channel-State: [CS_EXECUTE] Channel-Call-State: [ACTIVE] Channel-State-Number: [4] Channel-Name: [sofia/internal/229 at tssec.ru] Unique-ID: [1371631e-7abc-11df-97d4-6f7c82b8b4b4] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-Presence-ID: [229 at tssec.ru] Answer-State: [answered] Channel-Read-Codec-Name: [PCMA] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMA] Channel-Write-Codec-Rate: [8000] Caller-Username: [229] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Nikolai Gabelok] Caller-Caller-ID-Number: [229] Caller-Callee-ID-Name: [215] Caller-Callee-ID-Number: [215] Caller-Network-Addr: [192.168.0.176] Caller-ANI: [229] Caller-Destination-Number: [215] Caller-Unique-ID: [1371631e-7abc-11df-97d4-6f7c82b8b4b4] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/229 at tssec.ru] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1276853309465278] Caller-Channel-Created-Time: [1276853309465278] Caller-Channel-Answered-Time: [1276853311690473] Caller-Channel-Progress-Time: [1276853309605353] Caller-Channel-Progress-Media-Time: [1276853309481286] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] proto: [src/switch_channel.c] login: [src/switch_channel.c] from: [229 at tssec.ru] rpid: [unknown] status: [answered] event_type: [presence] alt_event_type: [dialog] presence-call-direction: [inbound] event_count: [2] 2010-06-18 13:28:31.782523 [NOTICE] sofia_presence.c:1126 SEND PRESENCE To: 100 at tssec.ru From: 229 at tssec.ru Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 Profile: internal [internal] 2010-06-18 13:28:31.782523 [INFO] sofia_presence.c:682 IN END_PRESENCE_SQL (internal) 2010-06-18 13:28:31.782523 [WARNING] sofia_presence.c:593 tssec.ru is an alias, skipping 2010-06-18 13:28:31.782523 [WARNING] sofia_presence.c:600 external is passive, skipping 2010-06-18 13:28:31.982630 [INFO] switch_rtp.c:2459 Auto Changing port from 10.8.6.6:16535 to 10.8.6.6:16534 recv 610 bytes from udp/[192.168.0.176]:5060 at 09:30:12.688760: ------------------------------------------------------------------------ BYE sip:215 at 192.168.50.11:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-15ec1600 From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" ;tag=D0rpvjgDaB86m Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="229",realm="tssec.ru",nonce="13679dc0-7abc-11df-97d3-6f7c82b8b4b4",uri="sip:215 at 192.168.50.11:5060",algorithm=MD5,response="4bc49e6ad1c9587cccbb4ec6cd8378d2",qop=auth,nc=00000002,cnonce="4faa6b95" User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ 2010-06-18 13:28:37.849779 [DEBUG] switch_channel.c:2257 (sofia/internal/229 at tssec.ru) Callstate Change ACTIVE -> HANGUP 2010-06-18 13:28:37.849779 [NOTICE] sofia.c:481 Hangup sofia/internal/229 at tssec.ru [CS_EXECUTE] [NORMAL_CLEARING] 2010-06-18 13:28:37.849779 [DEBUG] switch_channel.c:2273 Send signal sofia/internal/229 at tssec.ru [KILL] 2010-06-18 13:28:37.861786 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/229 at tssec.ru ending bridge by request from read function 2010-06-18 13:28:37.861786 [DEBUG] switch_ivr_bridge.c:472 sofia/internal/229 at tssec.ru ending bridge by request from write function 2010-06-18 13:28:37.861786 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD DONE [sofia/internal/229 at tssec.ru] 2010-06-18 13:28:37.861786 [DEBUG] switch_ivr_bridge.c:585 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:37.861786 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:37.861786 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD DONE [sofia/internal/sip:215 at 10.8.6.6:5080] 2010-06-18 13:28:37.861786 [DEBUG] switch_ivr_bridge.c:585 Send signal sofia/internal/229 at tssec.ru [BREAK] 2010-06-18 13:28:37.861786 [DEBUG] switch_channel.c:2257 (sofia/internal/sip:215 at 10.8.6.6:5080) Callstate Change ACTIVE -> HANGUP 2010-06-18 13:28:37.861786 [NOTICE] switch_ivr_bridge.c:637 Hangup sofia/internal/sip:215 at 10.8.6.6:5080 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2010-06-18 13:28:37.861786 [DEBUG] switch_channel.c:2273 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [KILL] 2010-06-18 13:28:37.861786 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:37.861786 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:215 at 10.8.6.6:5080) State EXCHANGE_MEDIA going to sleep 2010-06-18 13:28:37.861786 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 10.8.6.6:5080) Running State Change CS_HANGUP 2010-06-18 13:28:37.861786 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/sip:215 at 10.8.6.6:5080) State HANGUP 2010-06-18 13:28:37.861786 [DEBUG] mod_sofia.c:435 sofia/internal/sip:215 at 10.8.6.6:5080 Overriding SIP cause 480 with 200 from the other leg 2010-06-18 13:28:37.861786 [DEBUG] mod_sofia.c:441 Channel sofia/internal/sip:215 at 10.8.6.6:5080 hanging up, cause: NORMAL_CLEARING 2010-06-18 13:28:37.861786 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/229 at tssec.ru) State EXECUTE going to sleep 2010-06-18 13:28:37.861786 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/229 at tssec.ru) Running State Change CS_HANGUP 2010-06-18 13:28:37.861786 [DEBUG] switch_ivr_async.c:487 Stop recording file /var/lib/freeswitch/recordings/229.215.2010-06-18-13-28.wav 2010-06-18 13:28:37.869790 [DEBUG] switch_core_media_bug.c:418 Removing BUG from sofia/internal/229 at tssec.ru 2010-06-18 13:28:37.869790 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/229 at tssec.ru) State HANGUP 2010-06-18 13:28:37.877794 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/229 at tssec.ru [BREAK] send 490 bytes to udp/[192.168.0.176]:5060 at 09:30:12.716775: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK-15ec1600 From: "Nikolai Gabelok" ;tag=1e57eac0cc8bf959o0 To: "Elizarov Vladimir" ;tag=D0rpvjgDaB86m Call-ID: b4d6c80c-3143dc7d at 192.168.0.176 CSeq: 103 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2010-06-18 13:28:37.877794 [DEBUG] mod_sofia.c:435 sofia/internal/229 at tssec.ru Overriding SIP cause 480 with 200 from the other leg 2010-06-18 13:28:37.877794 [DEBUG] mod_sofia.c:441 Channel sofia/internal/229 at tssec.ru hanging up, cause: NORMAL_CLEARING 2010-06-18 13:28:37.885798 [DEBUG] mod_sofia.c:484 Sending BYE to sofia/internal/sip:215 at 10.8.6.6:5080 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:215 at 10.8.6.6:5080 Standard HANGUP, cause: NORMAL_CLEARING 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/sip:215 at 10.8.6.6:5080) State HANGUP going to sleep 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/sip:215 at 10.8.6.6:5080) State Change CS_HANGUP -> CS_REPORTING 2010-06-18 13:28:37.885798 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:215 at 10.8.6.6:5080) Running State Change CS_REPORTING 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/sip:215 at 10.8.6.6:5080) State REPORTING 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:215 at 10.8.6.6:5080 Standard REPORTING, cause: NORMAL_CLEARING 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/sip:215 at 10.8.6.6:5080) State REPORTING going to sleep 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/sip:215 at 10.8.6.6:5080) State Change CS_REPORTING -> CS_DESTROY 2010-06-18 13:28:37.885798 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/sip:215 at 10.8.6.6:5080 [BREAK] 2010-06-18 13:28:37.885798 [DEBUG] switch_core_session.c:1171 Session 115 (sofia/internal/sip:215 at 10.8.6.6:5080) Locked, Waiting on external entities 2010-06-18 13:28:37.885798 [NOTICE] switch_core_session.c:1189 Session 115 (sofia/internal/sip:215 at 10.8.6.6:5080) Ended 2010-06-18 13:28:37.885798 [NOTICE] switch_core_session.c:1191 Close Channel sofia/internal/sip:215 at 10.8.6.6:5080 [CS_DESTROY] 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/sip:215 at 10.8.6.6:5080) Callstate Change HANGUP -> DOWN 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/sip:215 at 10.8.6.6:5080) Running State Change CS_DESTROY 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:215 at 10.8.6.6:5080) State DESTROY 2010-06-18 13:28:37.885798 [DEBUG] mod_sofia.c:350 sofia/internal/sip:215 at 10.8.6.6:5080 SOFIA DESTROY 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:215 at 10.8.6.6:5080 Standard DESTROY 2010-06-18 13:28:37.885798 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:215 at 10.8.6.6:5080) State DESTROY going to sleep send 612 bytes to udp/[10.8.6.6]:5080 at 09:30:12.724779: ------------------------------------------------------------------------ BYE sip:215 at 10.8.6.6:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.11;rport;branch=z9hG4bK8jSUvjrN60v4Q Max-Forwards: 70 From: "Nikolai Gabelok" ;tag=e9HFyD1g7KySg To: ;tag=fyiai Call-ID: ead3e978-f55e-122d-a480-00163efcbed2 CSeq: 132308367 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2010-06-18 13:28:37.885798 [INFO] sofia_presence.c:663 IN START_PRESENCE_SQL (internal) 2010-06-18 13:28:37.885798 [ERR] sofia_presence.c:672 DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Call Ended','unknown','tssec.ru',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='229' and (sub_to_host='tssec.ru' or presence_hosts like '%tssec.ru%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: Event-Name: [PRESENCE_IN] Core-UUID: [9341fc9e-7a4d-11df-92e6-6f7c82b8b4b4] FreeSWITCH-Hostname: [sip1.lan.tssec.ru] FreeSWITCH-IPv4: [192.168.50.11] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2010-06-18 13:28:37] Event-Date-GMT: [Fri, 18 Jun 2010 09:28:37 GMT] Event-Date-Timestamp: [1276853317861786] Event-Calling-File: [switch_channel.c] Event-Calling-Function: [switch_channel_presence] Event-Calling-Line-Number: [575] Channel-State: [CS_HANGUP] Channel-Call-State: [HANGUP] Channel-State-Number: [10] Channel-Name: [sofia/internal/229 at tssec.ru] Unique-ID: [1371631e-7abc-11df-97d4-6f7c82b8b4b4] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-Presence-ID: [229 at tssec.ru] Answer-State: [answered] Channel-Read-Codec-Name: [PCMA] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMA] Channel-Write-Codec-Rate: [8000] Caller-Username: [229] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Nikolai Gabelok] Caller-Caller-ID-Number: [229] Caller-Callee-ID-Name: [215] Caller-Callee-ID-Number: [215] Caller-Network-Addr: [192.168.0.176] Caller-ANI: [229] Caller-Destination-Number: [215] Caller-Unique-ID: [1371631e-7abc-11df-97d4-6f7c82b8b4b4] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/229 at tssec.ru] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1276853309465278] Caller-Channel-Created-Time: [1276853309465278] Caller-Channel-Answered-Time: [1276853311690473] Caller-Channel-Progress-Time: [1276853309605353] Caller-Channel-Progress-Media-Time: [1276853309481286] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [229] Other-Leg-Dialplan: [XML] Other-Leg-Caller-ID-Name: [Nikolai Gabelok] Other-Leg-Caller-ID-Number: [229] Other-Leg-Callee-ID-Name: [215] Other-Leg-Callee-ID-Number: [215] Other-Leg-Network-Addr: [10.8.6.6] Other-Leg-ANI: [229] Other-Leg-Destination-Number: [215] Other-Leg-Unique-ID: [1375a974-7abc-11df-97d5-6f7c82b8b4b4] Other-Leg-Source: [mod_sofia] Other-Leg-Context: [default] Other-Leg-Channel-Name: [sofia/internal/sip:215 at 10.8.6.6:5080] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] proto: [src/switch_channel.c] login: [src/switch_channel.c] from: [229 at tssec.ru] rpid: [unknown] status: [CS_HANGUP] event_type: [presence] alt_event_type: [dialog] presence-call-direction: [inbound] event_count: [3] 2010-06-18 13:28:37.889801 [NOTICE] sofia_presence.c:1126 SEND PRESENCE To: 100 at tssec.ru From: 229 at tssec.ru Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 Profile: internal [internal] 2010-06-18 13:28:37.889801 [WARNING] sofia_presence.c:1350 send payload: terminated 2010-06-18 13:28:37.889801 [INFO] sofia_presence.c:682 IN END_PRESENCE_SQL (internal) 2010-06-18 13:28:37.889801 [WARNING] sofia_presence.c:593 tssec.ru is an alias, skipping 2010-06-18 13:28:37.889801 [WARNING] sofia_presence.c:600 external is passive, skipping 2010-06-18 13:28:37.889801 [INFO] sofia_presence.c:663 IN START_PRESENCE_SQL (internal) 2010-06-18 13:28:37.889801 [ERR] sofia_presence.c:672 DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'NORMAL_CLEARING','unknown','tssec.ru',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='229' and (sub_to_host='tssec.ru' or presence_hosts like '%tssec.ru%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: Event-Name: [PRESENCE_IN] Core-UUID: [9341fc9e-7a4d-11df-92e6-6f7c82b8b4b4] FreeSWITCH-Hostname: [sip1.lan.tssec.ru] FreeSWITCH-IPv4: [192.168.50.11] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2010-06-18 13:28:37] Event-Date-GMT: [Fri, 18 Jun 2010 09:28:37 GMT] Event-Date-Timestamp: [1276853317869790] Event-Calling-File: [switch_channel.c] Event-Calling-Function: [switch_channel_presence] Event-Calling-Line-Number: [575] Channel-State: [CS_HANGUP] Channel-Call-State: [HANGUP] Channel-State-Number: [10] Channel-Name: [sofia/internal/229 at tssec.ru] Unique-ID: [1371631e-7abc-11df-97d4-6f7c82b8b4b4] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-Presence-ID: [229 at tssec.ru] Answer-State: [answered] Channel-Read-Codec-Name: [PCMA] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMA] Channel-Write-Codec-Rate: [8000] Caller-Username: [229] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Nikolai Gabelok] Caller-Caller-ID-Number: [229] Caller-Callee-ID-Name: [215] Caller-Callee-ID-Number: [215] Caller-Network-Addr: [192.168.0.176] Caller-ANI: [229] Caller-Destination-Number: [215] Caller-Unique-ID: [1371631e-7abc-11df-97d4-6f7c82b8b4b4] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/229 at tssec.ru] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1276853309465278] Caller-Channel-Created-Time: [1276853309465278] Caller-Channel-Answered-Time: [1276853311690473] Caller-Channel-Progress-Time: [1276853309605353] Caller-Channel-Progress-Media-Time: [1276853309481286] Caller-Channel-Hangup-Time: [1276853317861786] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [229] Other-Leg-Dialplan: [XML] Other-Leg-Caller-ID-Name: [Nikolai Gabelok] Other-Leg-Caller-ID-Number: [229] Other-Leg-Callee-ID-Name: [215] Other-Leg-Callee-ID-Number: [215] Other-Leg-Network-Addr: [10.8.6.6] Other-Leg-ANI: [229] Other-Leg-Destination-Number: [215] Other-Leg-Unique-ID: [1375a974-7abc-11df-97d5-6f7c82b8b4b4] Other-Leg-Source: [mod_sofia] Other-Leg-Context: [default] Other-Leg-Channel-Name: [sofia/internal/sip:215 at 10.8.6.6:5080] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] proto: [src/switch_channel.c] login: [src/switch_channel.c] from: [229 at tssec.ru] rpid: [unknown] status: [NORMAL_CLEARING] event_type: [presence] alt_event_type: [dialog] presence-call-direction: [inbound] event_count: [4] send 1044 bytes to udp/[192.168.0.220]:5060 at 09:30:12.728782: ------------------------------------------------------------------------ NOTIFY sip:100 at 192.168.0.220:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.11;rport;branch=z9hG4bK9UjmyD9r39jQK Max-Forwards: 70 From: ;tag=Be64rve6FSU1D To: ;tag=beeaf51cf0d364d9 Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 CSeq: 132308350 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1156 Content-Type: application/dialog-info+xml Content-Length: 254 terminated ------------------------------------------------------------------------ 2010-06-18 13:28:37.889801 [NOTICE] sofia_presence.c:1126 SEND PRESENCE To: 100 at tssec.ru From: 229 at tssec.ru Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 Profile: internal [internal] 2010-06-18 13:28:37.889801 [WARNING] sofia_presence.c:1350 send payload: terminated 2010-06-18 13:28:37.889801 [INFO] sofia_presence.c:682 IN END_PRESENCE_SQL (internal) 2010-06-18 13:28:37.889801 [WARNING] sofia_presence.c:593 tssec.ru is an alias, skipping 2010-06-18 13:28:37.889801 [WARNING] sofia_presence.c:600 external is passive, skipping 2010-06-18 13:28:37.897805 [DEBUG] switch_core_state_machine.c:46 sofia/internal/229 at tssec.ru Standard HANGUP, cause: NORMAL_CLEARING 2010-06-18 13:28:37.897805 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/229 at tssec.ru) State HANGUP going to sleep 2010-06-18 13:28:37.897805 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/229 at tssec.ru) State Change CS_HANGUP -> CS_REPORTING 2010-06-18 13:28:37.897805 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/229 at tssec.ru [BREAK] 2010-06-18 13:28:37.897805 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/229 at tssec.ru) Running State Change CS_REPORTING 2010-06-18 13:28:37.897805 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/229 at tssec.ru) State REPORTING 2010-06-18 13:28:37.897805 [DEBUG] switch_core_state_machine.c:53 sofia/internal/229 at tssec.ru Standard REPORTING, cause: NORMAL_CLEARING 2010-06-18 13:28:37.897805 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/229 at tssec.ru) State REPORTING going to sleep 2010-06-18 13:28:37.897805 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/229 at tssec.ru) State Change CS_REPORTING -> CS_DESTROY 2010-06-18 13:28:37.897805 [DEBUG] switch_core_session.c:1023 Send signal sofia/internal/229 at tssec.ru [BREAK] 2010-06-18 13:28:37.897805 [DEBUG] switch_core_session.c:1171 Session 114 (sofia/internal/229 at tssec.ru) Locked, Waiting on external entities 2010-06-18 13:28:37.897805 [NOTICE] switch_core_session.c:1189 Session 114 (sofia/internal/229 at tssec.ru) Ended 2010-06-18 13:28:37.897805 [NOTICE] switch_core_session.c:1191 Close Channel sofia/internal/229 at tssec.ru [CS_DESTROY] recv 285 bytes from udp/[192.168.0.220]:5060 at 09:30:12.740788: ------------------------------------------------------------------------ SIP/2.0 200 OK To: ;tag=beeaf51cf0d364d9 From: ;tag=Be64rve6FSU1D Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 CSeq: 132308350 NOTIFY Via: SIP/2.0/UDP 192.168.50.11;branch=z9hG4bK9UjmyD9r39jQK Server: Linksys/SPA962-6.1.3(a) Content-Length: 0 ------------------------------------------------------------------------ send 1044 bytes to udp/[192.168.0.220]:5060 at 09:30:12.740788: ------------------------------------------------------------------------ NOTIFY sip:100 at 192.168.0.220:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.11;rport;branch=z9hG4bKa5BD08Sv0j99e Max-Forwards: 70 From: ;tag=Be64rve6FSU1D To: ;tag=beeaf51cf0d364d9 Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 CSeq: 132308351 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1156 Content-Type: application/dialog-info+xml Content-Length: 254 terminated ------------------------------------------------------------------------ 2010-06-18 13:28:37.897805 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/229 at tssec.ru) Callstate Change HANGUP -> DOWN 2010-06-18 13:28:37.901807 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/229 at tssec.ru) Running State Change CS_DESTROY 2010-06-18 13:28:37.901807 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/229 at tssec.ru) State DESTROY 2010-06-18 13:28:37.901807 [DEBUG] mod_sofia.c:350 sofia/internal/229 at tssec.ru SOFIA DESTROY 2010-06-18 13:28:37.901807 [DEBUG] switch_core_state_machine.c:60 sofia/internal/229 at tssec.ru Standard DESTROY 2010-06-18 13:28:37.901807 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/229 at tssec.ru) State DESTROY going to sleep recv 285 bytes from udp/[192.168.0.220]:5060 at 09:30:12.752794: ------------------------------------------------------------------------ SIP/2.0 200 OK To: ;tag=beeaf51cf0d364d9 From: ;tag=Be64rve6FSU1D Call-ID: e2bc3098-8a58ff8d at 192.168.0.220 CSeq: 132308351 NOTIFY Via: SIP/2.0/UDP 192.168.50.11;branch=z9hG4bKa5BD08Sv0j99e Server: Linksys/SPA962-6.1.3(a) Content-Length: 0 ------------------------------------------------------------------------ recv 328 bytes from udp/[10.8.6.6]:5080 at 09:30:12.812827: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.11;received=192.168.50.11;rport=5060;branch=z9hG4bK8jSUvjrN60v4Q To: ;tag=fyiai From: "Nikolai Gabelok" ;tag=e9HFyD1g7KySg Call-ID: ead3e978-f55e-122d-a480-00163efcbed2 CSeq: 132308367 BYE Server: Twinkle/1.4.2 Content-Length: 0 ------------------------------------------------------------------------ On Fri, Jun 18, 2010 at 11:35 AM, Vladimir Elizarov wrote: > i'm change dialplan with call_limit: > ? > ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > ? ? ? ? data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > ? ? ? ? data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> > ? ? ? ? data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > ? ? ? > ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? > ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? data="{ignore_early_media=true}${sofia_contact(internal/${dialed_extension}@$${domain})}"/> > ? ? ? ? data="not_answer-${dialed_extension} XML default"/> > ? ? ? > ? ? > > then the state spa932 phone for incoming calls stopped working. Always > light green. > > On Thu, Jun 17, 2010 at 4:57 PM, Vladimir Elizarov > wrote: >> ?I found a bug in the presence of the linksys spa932 (configure this >> article: http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932). >> If during the >> conversation (lamp on) to the subscriber, someone called, the lamp on >> the panel goes into the state off. Who can verify whether he has >> reproduced the same problem? >> >> Unit key: fnc=blf+sd+cp;sub=110@$PROXY >> >> freeswitch 1.0.6 (git 10 06 2010) >> >> Linksys spa962: >> Software Version: ? ? ? 6.1.3(a) >> Hardware Version: ? ? ? 1.0.3(917f) >> >> Linksys spa932 >> Unit Enable: ? ?Yes ? ? Unit Online: ? ?Yes >> Subscribe Expires: ? ? ?600 ? ? Subscribe Retry Interval: ? ? ? 6 >> HW Version: ? ? 1.0.6 ? SW Version: ? ? 2.0.2 >> >> Configure image: >> http://img192.imageshack.us/img192/2320/linksysspa932.png >> >> -- >> Best regards, Vladimir Elizarov >> > > > > -- > Best regards, Vladimir Elizarov > -- Best regards, Vladimir Elizarov From vkozak at abisoft.spb.ru Fri Jun 18 05:29:17 2010 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Fri, 18 Jun 2010 16:29:17 +0400 Subject: [Freeswitch-users] show caller info Message-ID: hi everybody. I have one problem. I need to show in phone caller number without IP address or caller number + domain without IP address. I use api originate command with origination_caller_id_name and origination_caller_id_number parameters. bgapi originate {origination_caller_id_name=125 at 123.12.13.14,origination_caller_id_number=123 at 123.12.13.14}[origination_uuid=6daa7b7e-97e4-4790-827d-44ff4f40fd18]sofia/internal/sip:1009 at 172.26.10.65:61802;rinstance=437cf350c3a4546f &park() FS cuts value of origination_caller_id_name parameter (delete specified ip) - it's ok. And FS deletes specified ip from value of origination_caller_id_number parameter and adds FS-IP. how else can I show caller info? how can I show caller domain? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/b51307ca/attachment.html From mranga at gmail.com Fri Jun 18 06:53:45 2010 From: mranga at gmail.com (M. Ranganathan) Date: Fri, 18 Jun 2010 09:53:45 -0400 Subject: [Freeswitch-users] How to play a prompt to an ongoing conference? Message-ID: Hello, I would like to know how to play an automatic prompt to an ongoing freeSWITCH conference. The prompt is a wav file and I need to play the prompt at a specific time when the conference is in progress. What is the simplest way to do this? Thank you for any help in advance. Regards, Ranga -- M. Ranganathan From testa at voicetechnology.com.br Fri Jun 18 07:00:59 2010 From: testa at voicetechnology.com.br (Fernando Testa) Date: Fri, 18 Jun 2010 11:00:59 -0300 Subject: [Freeswitch-users] H.323 advice Message-ID: Hi FolkS! What h.323 is more stable, mod_opal or mod_h323? Is it near-production for simple call setup/teardown? Application is a conference server. Thank you. -- Fernando Gregianin Testa Voice Technology Ltda +55 11 35882166 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/ba60f858/attachment.html From steveayre at gmail.com Fri Jun 18 07:06:32 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Jun 2010 15:06:32 +0100 Subject: [Freeswitch-users] G729 Licenses In-Reply-To: References: Message-ID: Hi Brian, This is something I had wondered too. I think he meant in the case the media bug (e.g. recording) stops, is the license returned to the pool either a) immediately when the media bug stops or b) when the call is hung up, even though it isn't required for the remainder of the call -Steve > for #3, if both legs are g729 will the license be returned when the media bug is stopped or when the call is hungup? On 17 June 2010 13:50, Brian West wrote: > Yes. > > /b > > On Jun 17, 2010, at 7:46 AM, stephen at stephenjc wrote: > > > for #3, if both legs are g729 will the license be returned when the media > bug is stopped or when the call is hungup? > > > > > > Thanks, > > Stephen C > > -All of my email addresses go to the same place > > -Save Paper, think before you print > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/ceb36a8d/attachment.html From cmrienzo at gmail.com Fri Jun 18 07:08:41 2010 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Fri, 18 Jun 2010 10:08:41 -0400 Subject: [Freeswitch-users] Build the mod_unimrcp module In-Reply-To: References: Message-ID: mod_unimrcp makes FreeSWITCH an MRCP client so it can do TTS/ASR using an MRCP server. If you do not have (or want to use) an MRCP server, then this is not the module for you. Also, your configuration suggests you are using Nuance Speech Server, but are requesting a voice, awb, which Nuance doesn't have. On Fri, Jun 18, 2010 at 12:56 AM, Thangappan.M wrote: > Dear all, > > I am in the process of developing IVR. So just planned to convert > my application to handle the TTS voice engine which is supported by > FreeSWITCH. > Got the mod_unimrcp module which is used to recognize the speech > and synthesize the text to voice. > For building the mod_unimrcp modules done the following steps. > Uncomment the mod_unimrcp line in the modules.conf file in > the FreeSWITCH source > Given make mod_unimrcp-install command. > In modules.conf.xml uncomment the modue="mod_unimrcp"/> > Configured the following dial plan > name="unimrcp"> > > expression="^4922$"> > > > data="tts_engine=unimrcp:nuance5-mrcp1"/> > > --> > > > > > > > While making the call to 4922 got the following error in the > FreeSWITCH console. > > > > [INFO] mod_dialplan_xml.c:418 Processing thangappan->4922 in context default > [NOTICE] mod_dptools.c:717 Channel [sofia/internal/1012 at 192.168.1.222] has been answered > > [INFO] mod_unimrcp.c:1499 speech_handle: name = unimrcp, rate = 8000, speed = 0, samples = 160, voice = , engine = unimrcp, param = nuance5-mrcp1 > [INFO] mod_unimrcp.c:1502 voice = awb, rate = 8000 > [NOTICE] mrcp_client.c:549 Create MRCP Handle 0x8b02400 [nuance5-mrcp1] > > [INFO] mrcp_client_session.c:142 Create Channel 0x8b02400 > [INFO] mrcp_client_session.c:398 Receive App Request 0x8b02400 [2] > [NOTICE] rtsp_client.c:255 Create RTSP Handle 0x8b04408 > [INFO] mrcp_client.c:901 Add MRCP Handle 0x8b02400 > > [NOTICE] mrcp_client_session.c:718 Add Control Channel 0x8b02400 > [INFO] mrcp_client_session.c:420 Send Offer 0x8b02400 [c:0 a:1 v:0] > *[ERR] mod_unimrcp.c:965 (TTS-0) Timed out waiting for channel to be ready > > [ERR] switch_ivr_play_say.c:2104 Invalid TTS module!* > [NOTICE] switch_core_state_machine.c:185 sofia/internal/1012 at 192.168.1.222 has executed the last dialplan instruction, hanging up. > > [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1012 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING] > [NOTICE] switch_core_session.c:1179 Session 1 (sofia/internal/1012 at 192.168.1.222) Ended > > [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/1012 at 192.168.1.222 [CS_DESTROY] > [NOTICE] switch_channel.c:669 New Channel sofia/internal/1012 at 192.168.1.222 [49ed5afa-79f0-11df-b531-3553f3a65c3c] > > So need to find a solution for that. > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/14105412/attachment.html From brian at freeswitch.org Fri Jun 18 07:13:43 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Jun 2010 09:13:43 -0500 Subject: [Freeswitch-users] G729 Licenses In-Reply-To: References: Message-ID: <328EA16A-C489-4781-ADED-433B77B7691B@freeswitch.org> On Jun 18, 2010, at 9:06 AM, Steven Ayre wrote: > Hi Brian, > > This is something I had wondered too. > > I think he meant in the case the media bug (e.g. recording) stops, is the license returned to the pool either > a) immediately when the media bug stops It should when the recording is stopped. > or > b) when the call is hung up, even though it isn't required for the remainder of the call > > -Steve From anthony.minessale at gmail.com Fri Jun 18 07:14:11 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 09:14:11 -0500 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: I said leg timeout beats the group confirm timeouts group_confirm_cancel_timeout is a whole different variable, when you set that to true it will stop all the timeouts as soon as you reach group_confirm execution {group_confirm_cancel_timeout=true}[leg_timeout=10]sofia/foo/foo at bar.com On Fri, Jun 18, 2010 at 12:50 AM, lakshmanan ganapathy wrote: > Dear Antony, > > Also in the leg_timeout wiki > http://wiki.freeswitch.org/wiki/Variable_leg_timeout, it is stated as > follows > > "If you are using group confirm then you can cancel the timeout by using > the group_confirm_cancel_timeoutchannel variable." > > > > On Thu, Jun 17, 2010 at 8:22 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> no there is no way, besides making both timeouts longer. >> you could file a feature request/bounty to ask for a feature to stop the >> leg timer when you reach the confirm. >> >> >> On Thu, Jun 17, 2010 at 4:23 AM, Nagalenoj H. wrote: >> >>> Anthony, >>> But, then there is no use. Am I right? Usually, we'll use the >>> group_confirm_cancel_timeout only when we need to override the leg_timeout. >>> But it happens in reverse in this case., >>> >>> I've tried using the group_confirm_cancel_timeout along with call_timeout >>> and things happening similar like setting leg_timout. >>> >>> Then, tried without setting leg_timeout and call_timeout explicitly. >>> * In this case if the callee doesn't picks the call, it >>> disconnects the leg in 30 secs. >>> * If he answers the call and the script continues to execute, the >>> leg is disconnected in 60 secs. >>> >>> What I need to do is, when the callee picks the call the leg_timeout >>> should not be accounted more and the leg shouldn't be disconnected because >>> of leg_timeout after that. >>> >>> Any other way of doing this?! >>> >>> >>> >>> On Tue, Jun 15, 2010 at 10:53 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> leg timeout beats the group confirm timeouts >>>> >>>> >>>> On Tue, Jun 15, 2010 at 12:28 AM, Nagalenoj H. wrote: >>>> >>>>> Dear friends, >>>>> I've tried using the group_confirm_cancel_timeout channel variable. >>>>> I've written a testing script to get digits before bridging. But, it doesn't >>>>> seem to be working. >>>>> >>>>> My understanding after reading wiki is, >>>>> * When I dial [leg_timeout=10]user/1005, if he answers before >>>>> timeout and in the process of giving digits, then the call shouldn't be >>>>> disconnected after the leg_timeout secs (10 sec in the example). >>>>> >>>>> But, When I experiment it, the call is getting disconnected after 10 >>>>> seconds and it doesn't bother whether the callee has answered the >>>>> call(Started giving digits) or not answered at all. >>>>> >>>>> I've checked it with nc as follows, >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: set >>>>> execute-app-arg: group_confirm_key=exec >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: set >>>>> execute-app-arg: group_confirm_file=perl /root/confirm.pl >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: set >>>>> execute-app-arg: group_confirm_cancel_timeout=1 >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: bridge >>>>> execute-app-arg: [leg_timeout=10]user/1005 >>>>> >>>>> And here is the script, >>>>> >>>>> use freeswitch; >>>>> our $session; >>>>> my $digit; >>>>> >>>>> while(1) { >>>>> # Wait till response timeout for the first digit. >>>>> $digit = $session->getDigits(1, "", 10000); >>>>> freeswitch::consoleLog ("info","Digit>>".$digit."<<"); >>>>> >>>>> if (! $session->ready() ) { >>>>> freeswitch::consoleLog("info","Going to Exit\n"); >>>>> last; >>>>> } >>>>> if (defined $digit and $digit ne "" ) { >>>>> freeswitch::consoleLog("info","DTMF received: >>>>> $digit\n"); >>>>> if ($digit eq '#') { >>>>> return; >>>>> } >>>>> } >>>>> else { >>>>> freeswitch::consoleLog("info","Timeout\n"); >>>>> $session->hangup(); >>>>> } >>>>> } >>>>> 1; >>>>> >>>>> If my understanding is right then, I believe there is something wrong >>>>> with channel_variable. >>>>> >>>>> Kindly help me to resolve this. >>>>> >>>>> -- >>>>> Regards, >>>>> Nagalenoj H. >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Regards, >>> Nagalenoj H. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100618/02e15420/attachment-0001.html From stephen at stephenjc.com Fri Jun 18 07:17:40 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Fri, 18 Jun 2010 10:17:40 -0400 Subject: [Freeswitch-users] G729 Licenses In-Reply-To: <328EA16A-C489-4781-ADED-433B77B7691B@freeswitch.org> References: <328EA16A-C489-4781-ADED-433B77B7691B@freeswitch.org> Message-ID: thanks, that is the answer i was looking for. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print On Fri, Jun 18, 2010 at 10:13 AM, Brian West wrote: > > On Jun 18, 2010, at 9:06 AM, Steven Ayre wrote: > > > Hi Brian, > > > > This is something I had wondered too. > > > > I think he meant in the case the media bug (e.g. recording) stops, is the > license returned to the pool either > > a) immediately when the media bug stops > > It should when the recording is stopped. > > > or > > b) when the call is hung up, even though it isn't required for the > remainder of the call > > > > -Steve > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/e53ca727/attachment.html From anthony.minessale at gmail.com Fri Jun 18 07:17:34 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 09:17:34 -0500 Subject: [Freeswitch-users] G729 Licenses In-Reply-To: References: Message-ID: A, immediately providing you have the latest mod_com_g729 and latest FreeSWITCH On Fri, Jun 18, 2010 at 9:06 AM, Steven Ayre wrote: > Hi Brian, > > This is something I had wondered too. > > I think he meant in the case the media bug (e.g. recording) stops, is the > license returned to the pool either > a) immediately when the media bug stops > or > b) when the call is hung up, even though it isn't required for the > remainder of the call > > -Steve > > > > > for #3, if both legs are g729 will the license be returned when the media > bug is stopped or when the call is hungup? > > On 17 June 2010 13:50, Brian West wrote: > >> Yes. >> >> /b >> >> On Jun 17, 2010, at 7:46 AM, stephen at stephenjc wrote: >> >> > for #3, if both legs are g729 will the license be returned when the >> media bug is stopped or when the call is hungup? >> > >> > >> > Thanks, >> > Stephen C >> > -All of my email addresses go to the same place >> > -Save Paper, think before you print >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100618/61628b02/attachment.html From anthony.minessale at gmail.com Fri Jun 18 07:20:40 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 09:20:40 -0500 Subject: [Freeswitch-users] How to play a prompt to an ongoing conference? In-Reply-To: References: Message-ID: conference play On Fri, Jun 18, 2010 at 8:53 AM, M. Ranganathan wrote: > Hello, > > I would like to know how to play an automatic prompt to an ongoing > freeSWITCH conference. The prompt is a wav file and I need to play > the prompt at a specific time when the conference is in progress. What > is the simplest way to do this? > > Thank you for any help in advance. > > Regards, > > Ranga > > -- > M. Ranganathan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100618/03fec11b/attachment.html From peter.olsson at visionutveckling.se Fri Jun 18 07:25:01 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 18 Jun 2010 16:25:01 +0200 Subject: [Freeswitch-users] H.323 advice In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C567D0BAAB7@cooper> I would say that mod_h323 works best, and for simple call scenarios it should work good enough. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Fernando Testa Skickat: den 18 juni 2010 16:01 Till: Freeswitch Users ?mne: [Freeswitch-users] H.323 advice Hi FolkS! What h.323 is more stable, mod_opal or mod_h323? Is it near-production for simple call setup/teardown? Application is a conference server. Thank you. -- Fernando Gregianin Testa Voice Technology Ltda +55 11 35882166 !DSPAM:4c1b7dc332931270011636! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/d055c6ff/attachment.html From diego.viola at gmail.com Fri Jun 18 08:14:07 2010 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 18 Jun 2010 12:14:07 -0300 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <201006171324.09460.sos@sokhapkin.dyndns.org> References: <1276734248530-5188763.post@n2.nabble.com> <201006171656.06384.errotan@elder.hu> <201006171324.09460.sos@sokhapkin.dyndns.org> Message-ID: What do you mean with "Sometime very buggy" ? -- I personally find FreeSWITCH to be very stable. Diego On Thu, Jun 17, 2010 at 2:24 PM, Sergey Okhapkin wrote: > It's a good idea. 911 at freeswitch.org SIP URI with $1 per minute rate :-D > > I want to thank you again for providing a really good software. Sometime > very > buggy, but overall it's very good. I will never return to asterisk. > > On Thursday 17 June 2010, Anthony Minessale wrote: > > I am going to guess 911 callcenter where Mike is bleeding somewhere and > > Erin is a police dispatcher. > > > > On Thu, Jun 17, 2010 at 9:56 AM, Pusk?s Zsolt wrote: > > > 2010. j?nius 17. 02.24.08 d?tummal benxmy az al?bbiakat ?rta: > > > > Hi, > > > > I'm quite new to freeswitch and voip, so this may be in some way a > noob > > > > question but I've dug through a lot of the freeswitch docs and done > > > > quite > > > > > > a > > > > > > > bit of searching and haven't figured it out yet. > > > > > > > > We're creating a relatively straightforward VoIP system to be used > > > > > > entirely > > > > > > > internally (ex: users can only connect with other registered users > > > > within our system) and we'd like to be able to combine combine two > > > > calls into a single audio stream to the user without the two calls > > > > hearing each other. For example, if I'm talking to Mike on line 1 and > > > > I'm talking to Erin on line 2, is there a way for me to hear both > Erin > > > > and Mike simultaneously > > > > > > but > > > > > > > for them not to hear each other? > > > > > > > > Alternatively, is there a straightforward way to simply merge the > calls > > > > into a conference-type experience where we all hear each other > without > > > > > > the > > > > > > > user explicitly setting up a conference call? > > > > > > > > Any and all input is greatly appreciated, as I'm up to my ears in > > > > freeswitch but have approximately zero experience with it! > > > > > > > > Ben > > > > > > Hi. > > > > > > What is the point of talking to 2 person while they can't hear each > other > > > ? For example when you say a sentence to person "A" and he replies back > > > with lots of sentences how person "B" knows when he can talk if he > can't > > > hear person "A"? Person "B" starts talking while person "A" so you > can't > > > understand > > > a word. This would only confuse people... > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.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/20100618/2010520e/attachment-0001.html From sos at sokhapkin.dyndns.org Fri Jun 18 08:28:28 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 18 Jun 2010 11:28:28 -0400 Subject: [Freeswitch-users] Merge two calls In-Reply-To: References: <1276734248530-5188763.post@n2.nabble.com> <201006171324.09460.sos@sokhapkin.dyndns.org> Message-ID: <201006181128.28407.sos@sokhapkin.dyndns.org> See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a workaround for MODSOFIA-59. On Friday 18 June 2010, Diego Viola wrote: > What do you mean with "Sometime very buggy" ? -- I personally find > FreeSWITCH to be very stable. > > Diego > > On Thu, Jun 17, 2010 at 2:24 PM, Sergey Okhapkin > > wrote: > > It's a good idea. 911 at freeswitch.org SIP URI with $1 per minute rate :-D > > > > I want to thank you again for providing a really good software. Sometime > > very > > buggy, but overall it's very good. I will never return to asterisk. > > > > On Thursday 17 June 2010, Anthony Minessale wrote: > > > I am going to guess 911 callcenter where Mike is bleeding somewhere and > > > Erin is a police dispatcher. > > > > > > On Thu, Jun 17, 2010 at 9:56 AM, Pusk?s Zsolt wrote: > > > > 2010. j?nius 17. 02.24.08 d?tummal benxmy az al?bbiakat ?rta: > > > > > Hi, > > > > > I'm quite new to freeswitch and voip, so this may be in some way a > > > > noob > > > > > > > question but I've dug through a lot of the freeswitch docs and done > > > > > quite > > > > > > > > a > > > > > > > > > bit of searching and haven't figured it out yet. > > > > > > > > > > We're creating a relatively straightforward VoIP system to be used > > > > > > > > entirely > > > > > > > > > internally (ex: users can only connect with other registered users > > > > > within our system) and we'd like to be able to combine combine two > > > > > calls into a single audio stream to the user without the two calls > > > > > hearing each other. For example, if I'm talking to Mike on line 1 > > > > > and I'm talking to Erin on line 2, is there a way for me to hear > > > > > both > > > > Erin > > > > > > > and Mike simultaneously > > > > > > > > but > > > > > > > > > for them not to hear each other? > > > > > > > > > > Alternatively, is there a straightforward way to simply merge the > > > > calls > > > > > > > into a conference-type experience where we all hear each other > > > > without > > > > > > the > > > > > > > > > user explicitly setting up a conference call? > > > > > > > > > > Any and all input is greatly appreciated, as I'm up to my ears in > > > > > freeswitch but have approximately zero experience with it! > > > > > > > > > > Ben > > > > > > > > Hi. > > > > > > > > What is the point of talking to 2 person while they can't hear each > > > > other > > > > > > ? For example when you say a sentence to person "A" and he replies > > > > back with lots of sentences how person "B" knows when he can talk if > > > > he > > > > can't > > > > > > hear person "A"? Person "B" starts talking while person "A" so you > > > > can't > > > > > > understand > > > > a word. This would only confuse people... > > > > > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.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 > From helmut.kuper at ewetel.de Fri Jun 18 08:42:25 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 18 Jun 2010 17:42:25 +0200 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: <4C19D4FD.1050800@ewetel.de> References: <4C12D0F7.5000300@ewetel.de> <4C13CAEC.3020005@ewetel.de> <4C19C9C9.1080806@ewetel.de> <4C19D4FD.1050800@ewetel.de> Message-ID: <4C1B93E1.7070700@ewetel.de> Hello, I found a solution which seems to works for me so far. At least I can use esl-phpmod now. Problem is, that php's socket_accept() functions returns a resource instead of a file descriptor. So esl-phpmod has to find out the corresponding file descriptor before "new ESLconnection()" is called. Here my svn diff for the FS ESL developer to get it reviewed: Index: php/esl_wrap.cpp =================================================================== --- php/esl_wrap.cpp (Revision 17782) +++ php/esl_wrap.cpp (Arbeitskopie) @@ -1793,7 +1793,7 @@ ZEND_NAMED_FUNCTION(_wrap_new_ESLconnection__SWIG_2) { - int arg1 ; + int arg1, type, *p; ESLconnection *result = 0 ; zval **args[1]; @@ -1806,6 +1806,12 @@ /*@SWIG:/usr/local/share/swig/1.3.35/php4/utils.i,7,CONVERT_INT_IN@*/ convert_to_long_ex(args[0]); arg1 = (int) Z_LVAL_PP(args[0]); + //Find the needed numeric file descriptor + //First get the Resource object + p=(int*)zend_list_find(arg1, &type); + if (!p) goto fail; + //Second, get the file descriptor + arg1 = *p; /*@SWIG@*/; result = (ESLconnection *)new ESLconnection(arg1); @@ -1825,9 +1831,7 @@ argc = ZEND_NUM_ARGS(); zend_get_parameters_array_ex(argc,argv); if (argc == 1) { - int _v; - _v = (Z_TYPE_PP(argv[0]) == IS_LONG); - if (_v) { + if (Z_TYPE_PP(argv[0]) == IS_RESOURCE) { return _wrap_new_ESLconnection__SWIG_2(INTERNAL_FUNCTION_PARAM_PASSTHRU); } } regards Helmut On 17.06.2010 09:55, Helmut Kuper wrote: > Hi, > > a bit success now. When I use this command: > > $con = new ESLconnection(intval($csock)); > > I got an ESLconnection object, but it is not usable. var_dump($con) shows > > object(ESLconnection)#1 (1) { > ["_cPtr"]=> > resource(7) of type (_p_ESLconnection) > } > > > which looks good to me, but strace still shows "Bad file descriptor": > > setsockopt(6, SOL_TCP, TCP_NODELAY, [1], 4) = -1 EBADF (Bad file descriptor) > sendto(6, "connect\n\n", 9, 0, NULL, 0) = -1 EBADF (Bad file descriptor) > > Calling $con->getINFO() results in NULL ... > > > regards > Helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Helmut Kuper Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Dr. Werner Brinker Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From helmut.kuper at ewetel.de Fri Jun 18 08:42:33 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 18 Jun 2010 17:42:33 +0200 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: <4C19D4FD.1050800@ewetel.de> References: <4C12D0F7.5000300@ewetel.de> <4C13CAEC.3020005@ewetel.de> <4C19C9C9.1080806@ewetel.de> <4C19D4FD.1050800@ewetel.de> Message-ID: <4C1B93E9.9020409@ewetel.de> Hello, I found a solution which seems to works for me so far. At least I can use esl-phpmod now. Problem is, that php's socket_accept() functions returns a resource instead of a file descriptor. So esl-phpmod has to find out the corresponding file descriptor before "new ESLconnection()" is called. Here my svn diff for the FS ESL developer to get it reviewed: Index: php/esl_wrap.cpp =================================================================== --- php/esl_wrap.cpp (Revision 17782) +++ php/esl_wrap.cpp (Arbeitskopie) @@ -1793,7 +1793,7 @@ ZEND_NAMED_FUNCTION(_wrap_new_ESLconnection__SWIG_2) { - int arg1 ; + int arg1, type, *p; ESLconnection *result = 0 ; zval **args[1]; @@ -1806,6 +1806,12 @@ /*@SWIG:/usr/local/share/swig/1.3.35/php4/utils.i,7,CONVERT_INT_IN@*/ convert_to_long_ex(args[0]); arg1 = (int) Z_LVAL_PP(args[0]); + //Find the needed numeric file descriptor + //First get the Resource object + p=(int*)zend_list_find(arg1, &type); + if (!p) goto fail; + //Second, get the file descriptor + arg1 = *p; /*@SWIG@*/; result = (ESLconnection *)new ESLconnection(arg1); @@ -1825,9 +1831,7 @@ argc = ZEND_NUM_ARGS(); zend_get_parameters_array_ex(argc,argv); if (argc == 1) { - int _v; - _v = (Z_TYPE_PP(argv[0]) == IS_LONG); - if (_v) { + if (Z_TYPE_PP(argv[0]) == IS_RESOURCE) { return _wrap_new_ESLconnection__SWIG_2(INTERNAL_FUNCTION_PARAM_PASSTHRU); } } regards Helmut On 17.06.2010 09:55, Helmut Kuper wrote: > Hi, > > a bit success now. When I use this command: > > $con = new ESLconnection(intval($csock)); > > I got an ESLconnection object, but it is not usable. var_dump($con) shows > > object(ESLconnection)#1 (1) { > ["_cPtr"]=> > resource(7) of type (_p_ESLconnection) > } > > > which looks good to me, but strace still shows "Bad file descriptor": > > setsockopt(6, SOL_TCP, TCP_NODELAY, [1], 4) = -1 EBADF (Bad file descriptor) > sendto(6, "connect\n\n", 9, 0, NULL, 0) = -1 EBADF (Bad file descriptor) > > Calling $con->getINFO() results in NULL ... > > > regards > Helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 macedoslm at gmail.com Fri Jun 18 09:07:16 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Fri, 18 Jun 2010 13:07:16 -0300 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: <4C1B93E1.7070700@ewetel.de> References: <4C12D0F7.5000300@ewetel.de> <4C13CAEC.3020005@ewetel.de> <4C19C9C9.1080806@ewetel.de> <4C19D4FD.1050800@ewetel.de> <4C1B93E1.7070700@ewetel.de> Message-ID: Hi Helmut, Thanks for the patch. I'll try it. Regards, -- Samuel Macedo On 18 June 2010 12:42, Helmut Kuper wrote: > Hello, > > I found a solution which seems to works for me so far. At least I can > use esl-phpmod now. > > Problem is, that php's socket_accept() functions returns a resource > instead of a file descriptor. So esl-phpmod has to find out the > corresponding file descriptor before "new ESLconnection()" is called. > > Here my svn diff for the FS ESL developer to get it reviewed: > > Index: php/esl_wrap.cpp > =================================================================== > --- php/esl_wrap.cpp (Revision 17782) > +++ php/esl_wrap.cpp (Arbeitskopie) > @@ -1793,7 +1793,7 @@ > > > ZEND_NAMED_FUNCTION(_wrap_new_ESLconnection__SWIG_2) { > - int arg1 ; > + int arg1, type, *p; > ESLconnection *result = 0 ; > zval **args[1]; > > @@ -1806,6 +1806,12 @@ > /*@SWIG:/usr/local/share/swig/1.3.35/php4/utils.i,7,CONVERT_INT_IN@*/ > convert_to_long_ex(args[0]); > arg1 = (int) Z_LVAL_PP(args[0]); > + //Find the needed numeric file descriptor > + //First get the Resource object > + p=(int*)zend_list_find(arg1, &type); > + if (!p) goto fail; > + //Second, get the file descriptor > + arg1 = *p; > /*@SWIG@*/; > > result = (ESLconnection *)new ESLconnection(arg1); > @@ -1825,9 +1831,7 @@ > argc = ZEND_NUM_ARGS(); > zend_get_parameters_array_ex(argc,argv); > if (argc == 1) { > - int _v; > - _v = (Z_TYPE_PP(argv[0]) == IS_LONG); > - if (_v) { > + if (Z_TYPE_PP(argv[0]) == IS_RESOURCE) { > return > _wrap_new_ESLconnection__SWIG_2(INTERNAL_FUNCTION_PARAM_PASSTHRU); > } > } > > > > regards > Helmut > > > On 17.06.2010 09:55, Helmut Kuper wrote: > > Hi, > > > > a bit success now. When I use this command: > > > > $con = new ESLconnection(intval($csock)); > > > > I got an ESLconnection object, but it is not usable. var_dump($con) shows > > > > object(ESLconnection)#1 (1) { > > ["_cPtr"]=> > > resource(7) of type (_p_ESLconnection) > > } > > > > > > which looks good to me, but strace still shows "Bad file descriptor": > > > > setsockopt(6, SOL_TCP, TCP_NODELAY, [1], 4) = -1 EBADF (Bad file > descriptor) > > sendto(6, "connect\n\n", 9, 0, NULL, 0) = -1 EBADF (Bad file descriptor) > > > > Calling $con->getINFO() results in NULL ... > > > > > > regards > > Helmut > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > Helmut Kuper > Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister > Telefax: (0441) 8000-2799 > mailto:helmut.kuper at ewetel.de > ___________________________________ > EWE TEL GmbH > Cloppenburger Stra?e 310 > 26133 Oldenburg > EWE TEL GmbH > > Handelsregister Amtsgericht Oldenburg HRB 3723 > Vorsitzender des Aufsichtsrates: Dr. Werner Brinker > Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), > Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole > Homepage: http://www.ewetel.de > ___________________________________ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/8cf3d392/attachment.html From erik.dekkers at wvds.nl Fri Jun 18 04:06:47 2010 From: erik.dekkers at wvds.nl (Erik Dekkers) Date: Fri, 18 Jun 2010 13:06:47 +0200 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: <7CFA8A39-97EB-4D46-986C-6891C555F351@freeswitch.org> References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> <1276803796935-5192528.post@n2.nabble.com> <00d701cb0e60$9cf23780$d6d6a680$@maly@molcs.org> <7CFA8A39-97EB-4D46-986C-6891C555F351@freeswitch.org> Message-ID: Brian, Can you tell what the Phone does wrong? Then we all can open open tickets, maybe it will be fixed afterall. Regards, Erik Dekkers (wvds-nl on #freeswitch) -----Oorspronkelijk bericht----- Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Brian West Verzonden: Thursday, June 17, 2010 11:11 PM Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? No it doesn't fully fix it.. again the phone DOES IT WRONG. We hacked FreeSWITCH to guess about it. Its still going to mess up. /b On Jun 17, 2010, at 4:04 PM, Mark Maly wrote: > All, > > Not smart enough to check to the "bit level" but received this from Aastra Tech support earlier and posted to another thread here, but ... > > "Your ticket number is xxxxx. > > We have just released a new GA firmware that I would like for you to load onto one of your phones to see if it resolves the issue. You can download this from our website at www.aastratelecom.com/support and click on Download Area and select the 6731i. The firmware version you are looking for is 2.6 and it is listed under Current Software Release. > > Please load that on a test phone and let us know the results. > > Thank you, > > Jessie Fetter > > Aastra Customer Technical Support > support at aastra.com > www.aastratelecom.com/support > 800-574-1611 > " > > It appears this fixed the problems I was encountering, but I'm not comfortable saying it'll always work. I don't really have a valid test plan, but SCA "appears" to work with the new firmware.... > > Mark _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 brian at freeswitch.org Fri Jun 18 09:31:37 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Jun 2010 11:31:37 -0500 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> <1276803796935-5192528.post@n2.nabble.com> <00d701cb0e60$9cf23780$d6d6a680$@maly@molcs.org> <7CFA8A39-97EB-4D46-986C-6891C555F351@freeswitch.org> Message-ID: Its missing the call_info header data to tell it what button the call transaction is associated with. /b On Jun 18, 2010, at 6:06 AM, Erik Dekkers wrote: > Brian, > > Can you tell what the Phone does wrong? Then we all can open open tickets, maybe it will be fixed afterall. > > Regards, > > Erik Dekkers > (wvds-nl on #freeswitch) From macedoslm at gmail.com Fri Jun 18 09:36:27 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Fri, 18 Jun 2010 13:36:27 -0300 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: References: <4C12D0F7.5000300@ewetel.de> <4C13CAEC.3020005@ewetel.de> <4C19C9C9.1080806@ewetel.de> <4C19D4FD.1050800@ewetel.de> <4C1B93E1.7070700@ewetel.de> Message-ID: Helmut, I've applied the patch and it's working well. Thanks for your help. Regards, -- Samuel Macedo On 18 June 2010 13:07, Samuel Macedo wrote: > Hi Helmut, > > Thanks for the patch. I'll try it. > > Regards, > -- > Samuel Macedo > > > On 18 June 2010 12:42, Helmut Kuper wrote: > >> Hello, >> >> I found a solution which seems to works for me so far. At least I can >> use esl-phpmod now. >> >> Problem is, that php's socket_accept() functions returns a resource >> instead of a file descriptor. So esl-phpmod has to find out the >> corresponding file descriptor before "new ESLconnection()" is called. >> >> Here my svn diff for the FS ESL developer to get it reviewed: >> >> Index: php/esl_wrap.cpp >> =================================================================== >> --- php/esl_wrap.cpp (Revision 17782) >> +++ php/esl_wrap.cpp (Arbeitskopie) >> @@ -1793,7 +1793,7 @@ >> >> >> ZEND_NAMED_FUNCTION(_wrap_new_ESLconnection__SWIG_2) { >> - int arg1 ; >> + int arg1, type, *p; >> ESLconnection *result = 0 ; >> zval **args[1]; >> >> @@ -1806,6 +1806,12 @@ >> /*@SWIG:/usr/local/share/swig/1.3.35/php4/utils.i,7,CONVERT_INT_IN@*/ >> convert_to_long_ex(args[0]); >> arg1 = (int) Z_LVAL_PP(args[0]); >> + //Find the needed numeric file descriptor >> + //First get the Resource object >> + p=(int*)zend_list_find(arg1, &type); >> + if (!p) goto fail; >> + //Second, get the file descriptor >> + arg1 = *p; >> /*@SWIG@*/; >> >> result = (ESLconnection *)new ESLconnection(arg1); >> @@ -1825,9 +1831,7 @@ >> argc = ZEND_NUM_ARGS(); >> zend_get_parameters_array_ex(argc,argv); >> if (argc == 1) { >> - int _v; >> - _v = (Z_TYPE_PP(argv[0]) == IS_LONG); >> - if (_v) { >> + if (Z_TYPE_PP(argv[0]) == IS_RESOURCE) { >> return >> _wrap_new_ESLconnection__SWIG_2(INTERNAL_FUNCTION_PARAM_PASSTHRU); >> } >> } >> >> >> >> regards >> Helmut >> >> >> On 17.06.2010 09:55, Helmut Kuper wrote: >> > Hi, >> > >> > a bit success now. When I use this command: >> > >> > $con = new ESLconnection(intval($csock)); >> > >> > I got an ESLconnection object, but it is not usable. var_dump($con) >> shows >> > >> > object(ESLconnection)#1 (1) { >> > ["_cPtr"]=> >> > resource(7) of type (_p_ESLconnection) >> > } >> > >> > >> > which looks good to me, but strace still shows "Bad file descriptor": >> > >> > setsockopt(6, SOL_TCP, TCP_NODELAY, [1], 4) = -1 EBADF (Bad file >> descriptor) >> > sendto(6, "connect\n\n", 9, 0, NULL, 0) = -1 EBADF (Bad file descriptor) >> > >> > Calling $con->getINFO() results in NULL ... >> > >> > >> > regards >> > Helmut >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 >> Helmut Kuper >> Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister >> Telefax: (0441) 8000-2799 >> mailto:helmut.kuper at ewetel.de >> ___________________________________ >> EWE TEL GmbH >> Cloppenburger Stra?e 310 >> 26133 Oldenburg >> EWE TEL GmbH >> >> Handelsregister Amtsgericht Oldenburg HRB 3723 >> Vorsitzender des Aufsichtsrates: Dr. Werner Brinker >> Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), >> Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole >> Homepage: http://www.ewetel.de >> ___________________________________ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/8e6477ce/attachment.html From anthony.minessale at gmail.com Fri Jun 18 10:08:23 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 12:08:23 -0500 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: References: <4C12D0F7.5000300@ewetel.de> <4C13CAEC.3020005@ewetel.de> <4C19C9C9.1080806@ewetel.de> <4C19D4FD.1050800@ewetel.de> <4C1B93E1.7070700@ewetel.de> Message-ID: That's unacceptable. you cannot patch a generated swig file. you need to learn what function in php gives you the raw file descriptor number from your socket and pass that to the constructor as required. The whole time your problem is that you don't know how to do this in PHP so you should stop looking in the code and just find the php method of a socket object to product the real file descriptor. On Fri, Jun 18, 2010 at 11:36 AM, Samuel Macedo wrote: > Helmut, > > I've applied the patch and it's working well. Thanks for your help. > > Regards, > -- > Samuel Macedo > > > On 18 June 2010 13:07, Samuel Macedo wrote: > >> Hi Helmut, >> >> Thanks for the patch. I'll try it. >> >> Regards, >> -- >> Samuel Macedo >> >> >> On 18 June 2010 12:42, Helmut Kuper wrote: >> >>> Hello, >>> >>> I found a solution which seems to works for me so far. At least I can >>> use esl-phpmod now. >>> >>> Problem is, that php's socket_accept() functions returns a resource >>> instead of a file descriptor. So esl-phpmod has to find out the >>> corresponding file descriptor before "new ESLconnection()" is called. >>> >>> Here my svn diff for the FS ESL developer to get it reviewed: >>> >>> Index: php/esl_wrap.cpp >>> =================================================================== >>> --- php/esl_wrap.cpp (Revision 17782) >>> +++ php/esl_wrap.cpp (Arbeitskopie) >>> @@ -1793,7 +1793,7 @@ >>> >>> >>> ZEND_NAMED_FUNCTION(_wrap_new_ESLconnection__SWIG_2) { >>> - int arg1 ; >>> + int arg1, type, *p; >>> ESLconnection *result = 0 ; >>> zval **args[1]; >>> >>> @@ -1806,6 +1806,12 @@ >>> /*@SWIG:/usr/local/share/swig/1.3.35/php4/utils.i,7,CONVERT_INT_IN@*/ >>> convert_to_long_ex(args[0]); >>> arg1 = (int) Z_LVAL_PP(args[0]); >>> + //Find the needed numeric file descriptor >>> + //First get the Resource object >>> + p=(int*)zend_list_find(arg1, &type); >>> + if (!p) goto fail; >>> + //Second, get the file descriptor >>> + arg1 = *p; >>> /*@SWIG@*/; >>> >>> result = (ESLconnection *)new ESLconnection(arg1); >>> @@ -1825,9 +1831,7 @@ >>> argc = ZEND_NUM_ARGS(); >>> zend_get_parameters_array_ex(argc,argv); >>> if (argc == 1) { >>> - int _v; >>> - _v = (Z_TYPE_PP(argv[0]) == IS_LONG); >>> - if (_v) { >>> + if (Z_TYPE_PP(argv[0]) == IS_RESOURCE) { >>> return >>> _wrap_new_ESLconnection__SWIG_2(INTERNAL_FUNCTION_PARAM_PASSTHRU); >>> } >>> } >>> >>> >>> >>> regards >>> Helmut >>> >>> >>> On 17.06.2010 09:55, Helmut Kuper wrote: >>> > Hi, >>> > >>> > a bit success now. When I use this command: >>> > >>> > $con = new ESLconnection(intval($csock)); >>> > >>> > I got an ESLconnection object, but it is not usable. var_dump($con) >>> shows >>> > >>> > object(ESLconnection)#1 (1) { >>> > ["_cPtr"]=> >>> > resource(7) of type (_p_ESLconnection) >>> > } >>> > >>> > >>> > which looks good to me, but strace still shows "Bad file descriptor": >>> > >>> > setsockopt(6, SOL_TCP, TCP_NODELAY, [1], 4) = -1 EBADF (Bad file >>> descriptor) >>> > sendto(6, "connect\n\n", 9, 0, NULL, 0) = -1 EBADF (Bad file >>> descriptor) >>> > >>> > Calling $con->getINFO() results in NULL ... >>> > >>> > >>> > regards >>> > Helmut >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at 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 >>> Helmut Kuper >>> Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister >>> Telefax: (0441) 8000-2799 >>> mailto:helmut.kuper at ewetel.de >>> ___________________________________ >>> EWE TEL GmbH >>> Cloppenburger Stra?e 310 >>> 26133 Oldenburg >>> EWE TEL GmbH >>> >>> Handelsregister Amtsgericht Oldenburg HRB 3723 >>> Vorsitzender des Aufsichtsrates: Dr. Werner Brinker >>> Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), >>> Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole >>> Homepage: http://www.ewetel.de >>> ___________________________________ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 > > -- 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/20100618/509b3c75/attachment-0001.html From robert.hadley at teotech.com Fri Jun 18 10:26:54 2010 From: robert.hadley at teotech.com (Robert Hadley) Date: Fri, 18 Jun 2010 10:26:54 -0700 Subject: [Freeswitch-users] Does FS need the pth (gnu pthreads) package? Message-ID: Does Freeswitch or the default enabled modules use or need the pthreads library, which on CentOS 5.3 is the pth package? Thanks, Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/c650b7a3/attachment.html From oseslija at gmail.com Fri Jun 18 10:50:13 2010 From: oseslija at gmail.com (Ognjen Seslija) Date: Fri, 18 Jun 2010 19:50:13 +0200 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <201006171324.09460.sos@sokhapkin.dyndns.org> References: <1276734248530-5188763.post@n2.nabble.com> <201006171656.06384.errotan@elder.hu> <201006171324.09460.sos@sokhapkin.dyndns.org> Message-ID: Not really good choice of words. "Very buggy" would indicate that the software keeps crashing and authors don't care at all. That is miles far from the real experience imho. On Thu, Jun 17, 2010 at 7:24 PM, Sergey Okhapkin wrote: > It's a good idea. 911 at freeswitch.org SIP URI with $1 per minute rate :-D > > I want to thank you again for providing a really good software. Sometime > very > buggy, but overall it's very good. I will never return to asterisk. > > On Thursday 17 June 2010, Anthony Minessale wrote: > > I am going to guess 911 callcenter where Mike is bleeding somewhere and > > Erin is a police dispatcher. > > > > On Thu, Jun 17, 2010 at 9:56 AM, Pusk?s Zsolt wrote: > > > 2010. j?nius 17. 02.24.08 d?tummal benxmy az al?bbiakat ?rta: > > > > Hi, > > > > I'm quite new to freeswitch and voip, so this may be in some way a > noob > > > > question but I've dug through a lot of the freeswitch docs and done > > > > quite > > > > > > a > > > > > > > bit of searching and haven't figured it out yet. > > > > > > > > We're creating a relatively straightforward VoIP system to be used > > > > > > entirely > > > > > > > internally (ex: users can only connect with other registered users > > > > within our system) and we'd like to be able to combine combine two > > > > calls into a single audio stream to the user without the two calls > > > > hearing each other. For example, if I'm talking to Mike on line 1 and > > > > I'm talking to Erin on line 2, is there a way for me to hear both > Erin > > > > and Mike simultaneously > > > > > > but > > > > > > > for them not to hear each other? > > > > > > > > Alternatively, is there a straightforward way to simply merge the > calls > > > > into a conference-type experience where we all hear each other > without > > > > > > the > > > > > > > user explicitly setting up a conference call? > > > > > > > > Any and all input is greatly appreciated, as I'm up to my ears in > > > > freeswitch but have approximately zero experience with it! > > > > > > > > Ben > > > > > > Hi. > > > > > > What is the point of talking to 2 person while they can't hear each > other > > > ? For example when you say a sentence to person "A" and he replies back > > > with lots of sentences how person "B" knows when he can talk if he > can't > > > hear person "A"? Person "B" starts talking while person "A" so you > can't > > > understand > > > a word. This would only confuse people... > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.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/20100618/acca9167/attachment.html From oseslija at gmail.com Fri Jun 18 10:53:22 2010 From: oseslija at gmail.com (Ognjen Seslija) Date: Fri, 18 Jun 2010 19:53:22 +0200 Subject: [Freeswitch-users] presence in linksys spa932 In-Reply-To: References: Message-ID: Hello, I never used a scenario where a watched phone has call waiting enabled, thus capable of receiving another call. I will try to reproduce this. Ognjen On Thu, Jun 17, 2010 at 2:57 PM, Vladimir Elizarov wrote: > I found a bug in the presence of the linksys spa932 (configure this > article: http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932). > If during the > conversation (lamp on) to the subscriber, someone called, the lamp on > the panel goes into the state off. Who can verify whether he has > reproduced the same problem? > > Unit key: fnc=blf+sd+cp;sub=110@$PROXY > > freeswitch 1.0.6 (git 10 06 2010) > > Linksys spa962: > Software Version: 6.1.3(a) > Hardware Version: 1.0.3(917f) > > Linksys spa932 > Unit Enable: Yes Unit Online: Yes > Subscribe Expires: 600 Subscribe Retry Interval: 6 > HW Version: 1.0.6 SW Version: 2.0.2 > > Configure image: > http://img192.imageshack.us/img192/2320/linksysspa932.png > > -- > Best regards, Vladimir Elizarov > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/45532a34/attachment.html From brian at freeswitch.org Fri Jun 18 11:08:32 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Jun 2010 13:08:32 -0500 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <201006181128.28407.sos@sokhapkin.dyndns.org> References: <1276734248530-5188763.post@n2.nabble.com> <201006171324.09460.sos@sokhapkin.dyndns.org> <201006181128.28407.sos@sokhapkin.dyndns.org> Message-ID: <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> Lets go over your bugs. MODSOFIA-72 - You have yet to provide the offending sip packet and you said it happens rarely. MODSOFIA-59 - HAS already been fixed have you not even tried the latest git checkout? We worked on this yesterday too. DP-16 - You'll have to hunt mikej or anthony down to talk about that one. If you like I can issue you a refund if the software is that awful. /b On Jun 18, 2010, at 10:28 AM, Sergey Okhapkin wrote: > See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a workaround for > MODSOFIA-59. From jesse at mactechs.com Fri Jun 18 11:07:31 2010 From: jesse at mactechs.com (Jesse Peterson) Date: Fri, 18 Jun 2010 11:07:31 -0700 Subject: [Freeswitch-users] Snom & SCA weirdness Message-ID: Hello, I've been playing with the SCA/SLA features ("manage-shared-appearance"/Broadsoft style) and have noticed some definite oddities with regard to our Snom phones (mostly 300's and 370's). Particularly how the buttons behave on the phones. I've tried different phone firmware (7.1.x, 7.3.x, 8.2.x, etc.) with all seemingly this same weird pattern: when a "Shared Line" button is selected that button's light turns on, but also the button LED two up gets turned on, too. E.g. If I had button 3 as a shared line, and I pushed it to make a call, LED 3 would light up but LED 1 would also light up. It appears as though the other phones sharing have a similar oddity where they'll very briefly light up the +2 LED but then it will go off and the correct shared-line button LED will turn on. This is with the FreeSwitch 1.6 release on a Mac OS X 10.6.4 machine. Does anybody have Snom & SCA/SLA working well and reliably? With which firmware and phones exactly? FreeSwitch trunk or release? Have there been improvements recently? Thanks! - Jesse P.S. Please Reply-All as I subscribe to the digest and may not immediately receive replies. Thank you. From sos at sokhapkin.dyndns.org Fri Jun 18 11:20:23 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 18 Jun 2010 14:20:23 -0400 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> References: <1276734248530-5188763.post@n2.nabble.com> <201006181128.28407.sos@sokhapkin.dyndns.org> <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> Message-ID: <201006181420.23860.sos@sokhapkin.dyndns.org> On Friday 18 June 2010, Brian West wrote: > Lets go over your bugs. > > MODSOFIA-72 - You have yet to provide the offending sip packet and you said > it happens rarely. The SIP packet is provided in the attachment to the bug. > MODSOFIA-59 - HAS already been fixed have you not even > tried the latest git checkout? We worked on this yesterday too. Good to know, will check right now. > > DP-16 - You'll have to hunt mikej or anthony down to talk about that one. > > If you like I can issue you a refund if the software is that awful. :-) > > /b > > On Jun 18, 2010, at 10:28 AM, Sergey Okhapkin wrote: > > See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a > > workaround for MODSOFIA-59. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 helmut.kuper at ewetel.de Fri Jun 18 11:55:51 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 18 Jun 2010 20:55:51 +0200 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: References: <4C12D0F7.5000300@ewetel.de> <4C13CAEC.3020005@ewetel.de> <4C19C9C9.1080806@ewetel.de> <4C19D4FD.1050800@ewetel.de> <4C1B93E1.7070700@ewetel.de> Message-ID: <4C1BC137.6040605@ewetel.de> Hi Anthony, yes, understand. Well, I've searched a few hours for a way to get it, but I found no - at least not in my centos standard php installation and not in Google. I found some hints (or requests) about dio_rawfd() there, but it seems such a function doesn't exists. If you or any other here have an idea where to look except adapting the code I would like to know it. You are right saying patching autogenerated code is a bad idea. But this is the only way I found so far. Maybe regards helmut Am 18.06.2010 19:08, schrieb Anthony Minessale: > That's unacceptable. you cannot patch a generated swig file. > > you need to learn what function in php gives you the raw file descriptor > number from your socket and pass that to the constructor as required. > > The whole time your problem is that you don't know how to do this in PHP > so you should stop looking in the code and just find the php method of a > socket object to product the real file descriptor. From sos at sokhapkin.dyndns.org Fri Jun 18 12:06:18 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 18 Jun 2010 15:06:18 -0400 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> References: <1276734248530-5188763.post@n2.nabble.com> <201006181128.28407.sos@sokhapkin.dyndns.org> <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> Message-ID: <201006181506.18118.sos@sokhapkin.dyndns.org> Just tried latest git, the issue MODSOFIA-59 is still there, caller gets wrong media port in SDP. On Friday 18 June 2010, Brian West wrote: > Lets go over your bugs. > > MODSOFIA-72 - You have yet to provide the offending sip packet and you said > it happens rarely. MODSOFIA-59 - HAS already been fixed have you not even > tried the latest git checkout? We worked on this yesterday too. > > DP-16 - You'll have to hunt mikej or anthony down to talk about that one. > > If you like I can issue you a refund if the software is that awful. > > /b > > On Jun 18, 2010, at 10:28 AM, Sergey Okhapkin wrote: > > See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a > > workaround for MODSOFIA-59. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 benxmy at gmail.com Fri Jun 18 12:12:33 2010 From: benxmy at gmail.com (benxmy) Date: Fri, 18 Jun 2010 12:12:33 -0700 (PDT) Subject: [Freeswitch-users] Merge two calls In-Reply-To: <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> References: <1276734248530-5188763.post@n2.nabble.com> <201006171656.06384.errotan@elder.hu> <201006171324.09460.sos@sokhapkin.dyndns.org> <201006181128.28407.sos@sokhapkin.dyndns.org> <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> Message-ID: <1276888353246-5196639.post@n2.nabble.com> It seems that the original subject of the thread pretty quickly disappeared... Does anyone have any suggestions involving a user listening to two independent lines at once while both those lines can hear the user but not each other? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Merge-two-calls-tp5188763p5196639.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Fri Jun 18 12:12:50 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 14:12:50 -0500 Subject: [Freeswitch-users] ESL, phpmod In-Reply-To: <4C1BC137.6040605@ewetel.de> References: <4C12D0F7.5000300@ewetel.de> <4C13CAEC.3020005@ewetel.de> <4C19C9C9.1080806@ewetel.de> <4C19D4FD.1050800@ewetel.de> <4C1B93E1.7070700@ewetel.de> <4C1BC137.6040605@ewetel.de> Message-ID: if php does not let you get the raw file descriptor from a socket, I will actually make an extra effort to make fun of them for being silly to take such an important thing away from a programmer. I am going to stick to the hope that they are not that foolish. maybe you can make your own binary php module to do it =p On Fri, Jun 18, 2010 at 1:55 PM, Helmut Kuper wrote: > Hi Anthony, > > yes, understand. Well, I've searched a few hours for a way to get it, > but I found no - at least not in my centos standard php installation and > not in Google. I found some hints (or requests) about dio_rawfd() there, > but it seems such a function doesn't exists. > > If you or any other here have an idea where to look except adapting the > code I would like to know it. You are right saying patching > autogenerated code is a bad idea. But this is the only way I found so > far. Maybe > > > regards > helmut > > Am 18.06.2010 19:08, schrieb Anthony Minessale: > > That's unacceptable. you cannot patch a generated swig file. > > > > you need to learn what function in php gives you the raw file descriptor > > number from your socket and pass that to the constructor as required. > > > > The whole time your problem is that you don't know how to do this in PHP > > so you should stop looking in the code and just find the php method of a > > socket object to product the real file descriptor. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100618/bd238b2c/attachment.html From anthony.minessale at gmail.com Fri Jun 18 12:20:05 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 14:20:05 -0500 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <201006181506.18118.sos@sokhapkin.dyndns.org> References: <1276734248530-5188763.post@n2.nabble.com> <201006181128.28407.sos@sokhapkin.dyndns.org> <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> <201006181506.18118.sos@sokhapkin.dyndns.org> Message-ID: 2 headsets On Fri, Jun 18, 2010 at 2:06 PM, Sergey Okhapkin wrote: > Just tried latest git, the issue MODSOFIA-59 is still there, caller gets > wrong > media port in SDP. > > On Friday 18 June 2010, Brian West wrote: > > Lets go over your bugs. > > > > MODSOFIA-72 - You have yet to provide the offending sip packet and you > said > > it happens rarely. MODSOFIA-59 - HAS already been fixed have you not > even > > tried the latest git checkout? We worked on this yesterday too. > > > > DP-16 - You'll have to hunt mikej or anthony down to talk about that one. > > > > If you like I can issue you a refund if the software is that awful. > > > > /b > > > > On Jun 18, 2010, at 10:28 AM, Sergey Okhapkin wrote: > > > See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a > > > workaround for MODSOFIA-59. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > -- 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/20100618/8f36cf0b/attachment.html From anthony.minessale at gmail.com Fri Jun 18 12:21:32 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 14:21:32 -0500 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <201006181506.18118.sos@sokhapkin.dyndns.org> References: <1276734248530-5188763.post@n2.nabble.com> <201006181128.28407.sos@sokhapkin.dyndns.org> <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> <201006181506.18118.sos@sokhapkin.dyndns.org> Message-ID: please stop trolling on this list or face moderation. On Fri, Jun 18, 2010 at 2:06 PM, Sergey Okhapkin wrote: > Just tried latest git, the issue MODSOFIA-59 is still there, caller gets > wrong > media port in SDP. > > On Friday 18 June 2010, Brian West wrote: > > Lets go over your bugs. > > > > MODSOFIA-72 - You have yet to provide the offending sip packet and you > said > > it happens rarely. MODSOFIA-59 - HAS already been fixed have you not > even > > tried the latest git checkout? We worked on this yesterday too. > > > > DP-16 - You'll have to hunt mikej or anthony down to talk about that one. > > > > If you like I can issue you a refund if the software is that awful. > > > > /b > > > > On Jun 18, 2010, at 10:28 AM, Sergey Okhapkin wrote: > > > See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a > > > workaround for MODSOFIA-59. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > -- 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/20100618/99860f6c/attachment.html From dswardstrom at remotelink.com Fri Jun 18 12:57:26 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Fri, 18 Jun 2010 14:57:26 -0500 (CDT) Subject: [Freeswitch-users] How to play a prompt to an ongoing conference? In-Reply-To: <365844971.240.1276890838856.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <1608933122.242.1276891046560.JavaMail.root@srvr12.remotelinkml.com> The following command is the right command: conference play In JavaScript, the following version of this commanhd should work: apiExecute("conference", confname+" play "+Filename); One word of warning. If you use play to a session, it normally waits until it is done. As I understand it, the above command does not wait. I had to know when it was done, but since I also had to play it to a channel, I decided that they would complete at the same time. There is a separate "conference <> stop" command. I haven't seen anything to suggest a way to determine if something is still playing or recording. I know recording takes up slots in the conference and I suspect that play would also. However, they aren't reported on in a list. Perhaps there should be another version of list, one that reports on the auxiliary channels. Regards, Paul David Swardstrom From kemen04 at gmail.com Fri Jun 18 11:22:50 2010 From: kemen04 at gmail.com (Travis Kemen) Date: Fri, 18 Jun 2010 13:22:50 -0500 Subject: [Freeswitch-users] presence in linksys spa932 In-Reply-To: References: Message-ID: I see the same issue with polycom and snom phones (although we don't use snom anymore). There is a bug report open http://jira.freeswitch.org/browse/FSCORE-598. The use of mod_limit works for phones that would normally only want one call at a time anyhow, but that solution does not work for switchboard phones because they may need to grab another incoming call even if they are on a call currently. Travis On Fri, Jun 18, 2010 at 12:53 PM, Ognjen Seslija wrote: > Hello, > > I never used a scenario where a watched phone has call waiting enabled, > thus capable of receiving another call. > I will try to reproduce this. > > Ognjen > > On Thu, Jun 17, 2010 at 2:57 PM, Vladimir Elizarov < > xengelpublicx at gmail.com> wrote: > >> I found a bug in the presence of the linksys spa932 (configure this >> article: http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932). >> If during the >> conversation (lamp on) to the subscriber, someone called, the lamp on >> the panel goes into the state off. Who can verify whether he has >> reproduced the same problem? >> >> Unit key: fnc=blf+sd+cp;sub=110@$PROXY >> >> freeswitch 1.0.6 (git 10 06 2010) >> >> Linksys spa962: >> Software Version: 6.1.3(a) >> Hardware Version: 1.0.3(917f) >> >> Linksys spa932 >> Unit Enable: Yes Unit Online: Yes >> Subscribe Expires: 600 Subscribe Retry Interval: 6 >> HW Version: 1.0.6 SW Version: 2.0.2 >> >> Configure image: >> http://img192.imageshack.us/img192/2320/linksysspa932.png >> >> -- >> Best regards, Vladimir Elizarov >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100618/6ed5179a/attachment.html From anthony.minessale at gmail.com Fri Jun 18 13:07:58 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 15:07:58 -0500 Subject: [Freeswitch-users] presence in linksys spa932 In-Reply-To: References: Message-ID: no there isn't that issue has been closed... On Fri, Jun 18, 2010 at 1:22 PM, Travis Kemen wrote: > I see the same issue with polycom and snom phones (although we don't use > snom anymore). There is a bug report open > http://jira.freeswitch.org/browse/FSCORE-598. The use of mod_limit works > for phones that would normally only want one call at a time anyhow, but that > solution does not work for switchboard phones because they may need to grab > another incoming call even if they are on a call currently. > > Travis > > > On Fri, Jun 18, 2010 at 12:53 PM, Ognjen Seslija wrote: > >> Hello, >> >> I never used a scenario where a watched phone has call waiting enabled, >> thus capable of receiving another call. >> I will try to reproduce this. >> >> Ognjen >> >> On Thu, Jun 17, 2010 at 2:57 PM, Vladimir Elizarov < >> xengelpublicx at gmail.com> wrote: >> >>> I found a bug in the presence of the linksys spa932 (configure this >>> article: http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932). >>> If during the >>> conversation (lamp on) to the subscriber, someone called, the lamp on >>> the panel goes into the state off. Who can verify whether he has >>> reproduced the same problem? >>> >>> Unit key: fnc=blf+sd+cp;sub=110@$PROXY >>> >>> freeswitch 1.0.6 (git 10 06 2010) >>> >>> Linksys spa962: >>> Software Version: 6.1.3(a) >>> Hardware Version: 1.0.3(917f) >>> >>> Linksys spa932 >>> Unit Enable: Yes Unit Online: Yes >>> Subscribe Expires: 600 Subscribe Retry Interval: 6 >>> HW Version: 1.0.6 SW Version: 2.0.2 >>> >>> Configure image: >>> http://img192.imageshack.us/img192/2320/linksysspa932.png >>> >>> -- >>> Best regards, Vladimir Elizarov >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100618/2da00a45/attachment.html From brian at freeswitch.org Fri Jun 18 13:14:45 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Jun 2010 15:14:45 -0500 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <201006181506.18118.sos@sokhapkin.dyndns.org> References: <1276734248530-5188763.post@n2.nabble.com> <201006181128.28407.sos@sokhapkin.dyndns.org> <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> <201006181506.18118.sos@sokhapkin.dyndns.org> Message-ID: MODSOFIA-59 is not our bug as far as I can tell. You are using bypass media.. I'm pretty sure we just pass those SDP's thru as we receive them from one side to the other. I double checked and that appears to be what is taking place so the bug is not in FreeSWITCH as it doesn't even allocate ports or rtp for bypass calls. /b On Jun 18, 2010, at 2:06 PM, Sergey Okhapkin wrote: > Just tried latest git, the issue MODSOFIA-59 is still there, caller gets wrong > media port in SDP. > > On Friday 18 June 2010, Brian West wrote: >> Lets go over your bugs. >> >> MODSOFIA-72 - You have yet to provide the offending sip packet and you said >> it happens rarely. MODSOFIA-59 - HAS already been fixed have you not even >> tried the latest git checkout? We worked on this yesterday too. >> >> DP-16 - You'll have to hunt mikej or anthony down to talk about that one. >> >> If you like I can issue you a refund if the software is that awful. >> >> /b From brian at freeswitch.org Fri Jun 18 13:17:46 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Jun 2010 15:17:46 -0500 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <201006181506.18118.sos@sokhapkin.dyndns.org> References: <1276734248530-5188763.post@n2.nabble.com> <201006181128.28407.sos@sokhapkin.dyndns.org> <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> <201006181506.18118.sos@sokhapkin.dyndns.org> Message-ID: <484966F1-7458-4CCB-9901-83D07DD55B2F@freeswitch.org> I see what you're doing... and I know why its doing it. Wish your descriptions and logs were as clear as when Anthony explained to me what exactly was going on. /b On Jun 18, 2010, at 2:06 PM, Sergey Okhapkin wrote: > Just tried latest git, the issue MODSOFIA-59 is still there, caller gets wrong > media port in SDP. > > On Friday 18 June 2010, Brian West wrote: >> Lets go over your bugs. >> >> MODSOFIA-72 - You have yet to provide the offending sip packet and you said >> it happens rarely. MODSOFIA-59 - HAS already been fixed have you not even >> tried the latest git checkout? We worked on this yesterday too. >> >> DP-16 - You'll have to hunt mikej or anthony down to talk about that one. >> >> If you like I can issue you a refund if the software is that awful. >> >> /b > From anthony.minessale at gmail.com Fri Jun 18 14:04:43 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 16:04:43 -0500 Subject: [Freeswitch-users] Merge two calls In-Reply-To: <1276888353246-5196639.post@n2.nabble.com> References: <1276734248530-5188763.post@n2.nabble.com> <201006171656.06384.errotan@elder.hu> <201006171324.09460.sos@sokhapkin.dyndns.org> <201006181128.28407.sos@sokhapkin.dyndns.org> <09D4012F-B4A9-432D-AE04-F425863358A5@freeswitch.org> <1276888353246-5196639.post@n2.nabble.com> Message-ID: also you could put them in a conference and use the relationships to make them not able to hear each other. On Fri, Jun 18, 2010 at 2:12 PM, benxmy wrote: > > It seems that the original subject of the thread pretty quickly > disappeared... > > Does anyone have any suggestions involving a user listening to two > independent lines at once while both those lines can hear the user but not > each other? > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Merge-two-calls-tp5188763p5196639.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100618/cd1322d4/attachment-0001.html From anthony.minessale at gmail.com Fri Jun 18 14:13:52 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 16:13:52 -0500 Subject: [Freeswitch-users] Snom & SCA weirdness In-Reply-To: References: Message-ID: you might want to get a filter instead and stop using digest mode. snom and SCA is broken on all firmwares, you need to ask them to fix it. We have reported it to them already. SCA works reliably only on Polycom and Linksys On Fri, Jun 18, 2010 at 1:07 PM, Jesse Peterson wrote: > Hello, > > I've been playing with the SCA/SLA features > ("manage-shared-appearance"/Broadsoft style) and have noticed some definite > oddities with regard to our Snom phones (mostly 300's and 370's). > > Particularly how the buttons behave on the phones. I've tried different > phone firmware (7.1.x, 7.3.x, 8.2.x, etc.) with all seemingly this same > weird pattern: when a "Shared Line" button is selected that button's light > turns on, but also the button LED two up gets turned on, too. E.g. If I had > button 3 as a shared line, and I pushed it to make a call, LED 3 would light > up but LED 1 would also light up. It appears as though the other phones > sharing have a similar oddity where they'll very briefly light up the +2 LED > but then it will go off and the correct shared-line button LED will turn on. > This is with the FreeSwitch 1.6 release on a Mac OS X 10.6.4 machine. > > Does anybody have Snom & SCA/SLA working well and reliably? With which > firmware and phones exactly? FreeSwitch trunk or release? Have there been > improvements recently? > > Thanks! > - Jesse > > > P.S. Please Reply-All as I subscribe to the digest and may not immediately > receive replies. Thank you. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100618/f0620291/attachment.html From anthony.minessale at gmail.com Fri Jun 18 14:23:22 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Jun 2010 16:23:22 -0500 Subject: [Freeswitch-users] Does FS need the pth (gnu pthreads) package? In-Reply-To: References: Message-ID: on FS for linux many things need pthreads. On Fri, Jun 18, 2010 at 12:26 PM, Robert Hadley wrote: > > > Does Freeswitch or the default enabled modules use or need the pthreads > library, which on CentOS 5.3 is the pth package? > > > > Thanks, > > Robert > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100618/ad6ce251/attachment.html From wespear at gmail.com Fri Jun 18 15:19:16 2010 From: wespear at gmail.com (Wes Pearce) Date: Fri, 18 Jun 2010 15:19:16 -0700 (PDT) Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> <1276803796935-5192528.post@n2.nabble.com> <00d701cb0e60$9cf23780$d6d6a680$@maly@molcs.org> <7CFA8A39-97EB-4D46-986C-6891C555F351@freeswitch.org> Message-ID: <1276899556015-5197151.post@n2.nabble.com> Yeah, you can see in the trace above, the outgoing calls have headers like... Call-Info: ;appearance-index=1;appearance-state=active Call-Info: ;appearance-index=*;appearance-state=idle The incoming call only has... Call-Info: ;appearance-index=*;appearance-state=idle This is what we're talking about right? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-tp5191927p5197151.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Fri Jun 18 15:27:12 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Jun 2010 17:27:12 -0500 Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? In-Reply-To: <1276899556015-5197151.post@n2.nabble.com> References: <1276795957115-5191927.post@n2.nabble.com> <3B90D2A6-5C0E-4F89-8E77-6AAEF80A4865@freeswitch.org> <1903CD8F-A5CF-450C-B865-D0EE65D36977@freeswitch.org> <1276803796935-5192528.post@n2.nabble.com> <00d701cb0e60$9cf23780$d6d6a680$@maly@molcs.org> <7CFA8A39-97EB-4D46-986C-6891C555F351@freeswitch.org> <1276899556015-5197151.post@n2.nabble.com> Message-ID: Yep because their 180 ringing doesn't contain the info . ;) /b On Jun 18, 2010, at 5:19 PM, Wes Pearce wrote: > > Yeah, you can see in the trace above, the outgoing calls have headers like... > > Call-Info: ;appearance-index=1;appearance-state=active > Call-Info: ;appearance-index=*;appearance-state=idle > > The incoming call only has... > > Call-Info: ;appearance-index=*;appearance-state=idle > > This is what we're talking about right? > -- From jesse at mactechs.com Fri Jun 18 15:28:52 2010 From: jesse at mactechs.com (Jesse Peterson) Date: Fri, 18 Jun 2010 15:28:52 -0700 Subject: [Freeswitch-users] Snom & SCA weirdness In-Reply-To: References: Message-ID: <2BE0088F-1A1F-416F-8541-B2024C5B524F@mactechs.com> Gotcha. Looks like I'll have to investigate the "emulation" people have spoken about with the Snom button features. I've also updated this page: http://wiki.freeswitch.org/wiki/Shared_Line_Appearance Have you heard anything from Snom to date? Also I prefer digest mode. Thanks for the suggestion, though. ;) Thanks, - Jesse On Jun 18, 2010, at 2:13 PM, Anthony Minessale wrote: > you might want to get a filter instead and stop using digest mode. > > snom and SCA is broken on all firmwares, you need to ask them to fix it. > We have reported it to them already. > > SCA works reliably only on Polycom and Linksys > > On Fri, Jun 18, 2010 at 1:07 PM, Jesse Peterson wrote: > Hello, > > I've been playing with the SCA/SLA features ("manage-shared-appearance"/Broadsoft style) and have noticed some definite oddities with regard to our Snom phones (mostly 300's and 370's). > > Particularly how the buttons behave on the phones. I've tried different phone firmware (7.1.x, 7.3.x, 8.2.x, etc.) with all seemingly this same weird pattern: when a "Shared Line" button is selected that button's light turns on, but also the button LED two up gets turned on, too. E.g. If I had button 3 as a shared line, and I pushed it to make a call, LED 3 would light up but LED 1 would also light up. It appears as though the other phones sharing have a similar oddity where they'll very briefly light up the +2 LED but then it will go off and the correct shared-line button LED will turn on. This is with the FreeSwitch 1.6 release on a Mac OS X 10.6.4 machine. > > Does anybody have Snom & SCA/SLA working well and reliably? With which firmware and phones exactly? FreeSwitch trunk or release? Have there been improvements recently? > From infos at madovsky.org Fri Jun 18 16:07:17 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 18 Jun 2010 19:07:17 -0400 Subject: [Freeswitch-users] question about NibbleBill Message-ID: <61FDF212ED8B4D42AE335E4211226EFE@MOBILEE1705> hi there, I'm trying to create user who can call for free and other not. I set nobal_action in nibblebill conf to an extension in my default dialplan but if the user is at $0 for special case I would authorize the call for free. is it possible ? Thanks F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/4188e237/attachment.html From sos at sokhapkin.dyndns.org Fri Jun 18 16:19:20 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 18 Jun 2010 19:19:20 -0400 Subject: [Freeswitch-users] question about NibbleBill In-Reply-To: <61FDF212ED8B4D42AE335E4211226EFE@MOBILEE1705> References: <61FDF212ED8B4D42AE335E4211226EFE@MOBILEE1705> Message-ID: <201006181919.20344.sos@sokhapkin.dyndns.org> Just do not enable nibbling for that user, do not set nibble_account variable. On Friday 18 June 2010, Madovsky wrote: > hi there, > > I'm trying to create user who can call for free and other not. > I set nobal_action in nibblebill conf to an extension in my default > dialplan but if the user is at $0 for special case I would authorize the > call for free. is it possible ? > > Thanks > > F > From infos at madovsky.org Fri Jun 18 16:21:33 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 18 Jun 2010 19:21:33 -0400 Subject: [Freeswitch-users] question about NibbleBill Message-ID: <198C973F6B1E41F49366943933FFC1DF@MOBILEE1705> sorry forget my question. I changed the order of the extensions in the default dialplan and set and seems to work Thx F ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Friday, June 18, 2010 7:07 PM Subject: question about NibbleBill hi there, I'm trying to create user who can call for free and other not. I set nobal_action in nibblebill conf to an extension in my default dialplan but if the user is at $0 for special case I would authorize the call for free. is it possible ? Thanks F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/7050e819/attachment-0001.html From thangappan143 at gmail.com Fri Jun 18 22:28:20 2010 From: thangappan143 at gmail.com (Thangappan.M) Date: Sat, 19 Jun 2010 10:58:20 +0530 Subject: [Freeswitch-users] Build the mod_unimrcp module In-Reply-To: References: Message-ID: Where can I get the speech synthesizers? How do I know the voices which are supported by particular servers? On Fri, Jun 18, 2010 at 7:38 PM, Christopher Rienzo wrote: > mod_unimrcp makes FreeSWITCH an MRCP client so it can do TTS/ASR using an > MRCP server. If you do not have (or want to use) an MRCP server, then this > is not the module for you. > > Also, your configuration suggests you are using Nuance Speech Server, but > are requesting a voice, awb, which Nuance doesn't have. > > > On Fri, Jun 18, 2010 at 12:56 AM, Thangappan.M wrote: > >> Dear all, >> >> I am in the process of developing IVR. So just planned to >> convert my application to handle the TTS voice engine which is supported by >> FreeSWITCH. >> Got the mod_unimrcp module which is used to recognize the speech >> and synthesize the text to voice. >> For building the mod_unimrcp modules done the following steps. >> Uncomment the mod_unimrcp line in the modules.conf file in >> the FreeSWITCH source >> Given make mod_unimrcp-install command. >> In modules.conf.xml uncomment the > modue="mod_unimrcp"/> >> Configured the following dial plan >> > name="unimrcp"> >> >> > expression="^4922$"> >> >> >> > data="tts_engine=unimrcp:nuance5-mrcp1"/> >> >> --> >> >> >> >> >> >> >> While making the call to 4922 got the following error in the >> FreeSWITCH console. >> >> >> >> [INFO] mod_dialplan_xml.c:418 Processing thangappan->4922 in context default >> >> [NOTICE] mod_dptools.c:717 Channel [sofia/internal/1012 at 192.168.1.222] has been answered >> >> [INFO] mod_unimrcp.c:1499 speech_handle: name = unimrcp, rate = 8000, speed = 0, samples = 160, voice = , engine = unimrcp, param = nuance5-mrcp1 >> [INFO] mod_unimrcp.c:1502 voice = awb, rate = 8000 >> [NOTICE] mrcp_client.c:549 Create MRCP Handle 0x8b02400 [nuance5-mrcp1] >> >> >> [INFO] mrcp_client_session.c:142 Create Channel 0x8b02400 >> [INFO] mrcp_client_session.c:398 Receive App Request 0x8b02400 [2] >> [NOTICE] rtsp_client.c:255 Create RTSP Handle 0x8b04408 >> >> [INFO] mrcp_client.c:901 Add MRCP Handle 0x8b02400 >> >> [NOTICE] mrcp_client_session.c:718 Add Control Channel 0x8b02400 >> [INFO] mrcp_client_session.c:420 Send Offer 0x8b02400 [c:0 a:1 v:0] >> *[ERR] mod_unimrcp.c:965 (TTS-0) Timed out waiting for channel to be ready >> >> >> [ERR] switch_ivr_play_say.c:2104 Invalid TTS module!* >> [NOTICE] switch_core_state_machine.c:185 sofia/internal/1012 at 192.168.1.222 has executed the last dialplan instruction, hanging up. >> >> >> [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1012 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING] >> [NOTICE] switch_core_session.c:1179 Session 1 (sofia/internal/1012 at 192.168.1.222) Ended >> >> >> [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/1012 at 192.168.1.222 [CS_DESTROY] >> [NOTICE] switch_channel.c:669 New Channel sofia/internal/1012 at 192.168.1.222 [49ed5afa-79f0-11df-b531-3553f3a65c3c] >> >> >> So need to find a solution for that. >> >> >> -- >> Regards, >> Thangappan.M >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100619/34ce0942/attachment.html From lakindia89 at gmail.com Fri Jun 18 22:50:14 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Sat, 19 Jun 2010 11:20:14 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: Can you please tell me what is meant by group confirm timeouts?? Here I've tested the group_confirm_cancel_timeout, and it didn't worked. But the same solution has worked for Phillip Jones Refer http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg14207.html. My scenario is, On Fri, Jun 18, 2010 at 7:44 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I said leg timeout beats the group confirm timeouts > > group_confirm_cancel_timeout is a whole different variable, when you set > that to true it will stop all the timeouts as soon as you reach > group_confirm execution > > {group_confirm_cancel_timeout=true}[leg_timeout=10]sofia/foo/foo at bar.com > > > > On Fri, Jun 18, 2010 at 12:50 AM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> Dear Antony, >> >> Also in the leg_timeout wiki >> http://wiki.freeswitch.org/wiki/Variable_leg_timeout, it is stated as >> follows >> >> "If you are using group confirm then you can cancel the timeout by using >> the group_confirm_cancel_timeoutchannel variable." >> >> >> >> On Thu, Jun 17, 2010 at 8:22 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> no there is no way, besides making both timeouts longer. >>> you could file a feature request/bounty to ask for a feature to stop the >>> leg timer when you reach the confirm. >>> >>> >>> On Thu, Jun 17, 2010 at 4:23 AM, Nagalenoj H. wrote: >>> >>>> Anthony, >>>> But, then there is no use. Am I right? Usually, we'll use the >>>> group_confirm_cancel_timeout only when we need to override the leg_timeout. >>>> But it happens in reverse in this case., >>>> >>>> I've tried using the group_confirm_cancel_timeout along with >>>> call_timeout and things happening similar like setting leg_timout. >>>> >>>> Then, tried without setting leg_timeout and call_timeout explicitly. >>>> * In this case if the callee doesn't picks the call, it >>>> disconnects the leg in 30 secs. >>>> * If he answers the call and the script continues to execute, >>>> the leg is disconnected in 60 secs. >>>> >>>> What I need to do is, when the callee picks the call the leg_timeout >>>> should not be accounted more and the leg shouldn't be disconnected because >>>> of leg_timeout after that. >>>> >>>> Any other way of doing this?! >>>> >>>> >>>> >>>> On Tue, Jun 15, 2010 at 10:53 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> leg timeout beats the group confirm timeouts >>>>> >>>>> >>>>> On Tue, Jun 15, 2010 at 12:28 AM, Nagalenoj H. wrote: >>>>> >>>>>> Dear friends, >>>>>> I've tried using the group_confirm_cancel_timeout channel >>>>>> variable. I've written a testing script to get digits before bridging. But, >>>>>> it doesn't seem to be working. >>>>>> >>>>>> My understanding after reading wiki is, >>>>>> * When I dial [leg_timeout=10]user/1005, if he answers before >>>>>> timeout and in the process of giving digits, then the call shouldn't be >>>>>> disconnected after the leg_timeout secs (10 sec in the example). >>>>>> >>>>>> But, When I experiment it, the call is getting disconnected after 10 >>>>>> seconds and it doesn't bother whether the callee has answered the >>>>>> call(Started giving digits) or not answered at all. >>>>>> >>>>>> I've checked it with nc as follows, >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_key=exec >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_file=perl /root/confirm.pl >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_cancel_timeout=1 >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: bridge >>>>>> execute-app-arg: [leg_timeout=10]user/1005 >>>>>> >>>>>> And here is the script, >>>>>> >>>>>> use freeswitch; >>>>>> our $session; >>>>>> my $digit; >>>>>> >>>>>> while(1) { >>>>>> # Wait till response timeout for the first digit. >>>>>> $digit = $session->getDigits(1, "", 10000); >>>>>> freeswitch::consoleLog ("info","Digit>>".$digit."<<"); >>>>>> >>>>>> if (! $session->ready() ) { >>>>>> freeswitch::consoleLog("info","Going to Exit\n"); >>>>>> last; >>>>>> } >>>>>> if (defined $digit and $digit ne "" ) { >>>>>> freeswitch::consoleLog("info","DTMF received: >>>>>> $digit\n"); >>>>>> if ($digit eq '#') { >>>>>> return; >>>>>> } >>>>>> } >>>>>> else { >>>>>> freeswitch::consoleLog("info","Timeout\n"); >>>>>> $session->hangup(); >>>>>> } >>>>>> } >>>>>> 1; >>>>>> >>>>>> If my understanding is right then, I believe there is something wrong >>>>>> with channel_variable. >>>>>> >>>>>> Kindly help me to resolve this. >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> Nagalenoj H. >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> Nagalenoj H. >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > > -- > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100619/bcb6799b/attachment-0001.html From lakindia89 at gmail.com Fri Jun 18 23:08:44 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Sat, 19 Jun 2010 11:38:44 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: Can you please tell me what is meant by group confirm timeouts?, because I thought group confirm timeouts means you are talking about group_confirm_cancel_timeout only. I'm sorry if my understanding is wrong. Here I've tested the group_confirm_cancel_timeout, and it didn't worked. But the same solution has worked for Phillip Jones Refer http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg14207.html. My scenario is, When a caller calls, an ESL Outbound socket program will be executed. The program will set group_confirm_key=exec and group_confirm_file=perl script.pl In the script.pl I'll get some digits from the user to get the confirmation, if true I'll allow him to bridge with the caller. I've also used [leg_timeout=10]. What happens is, when the user answers the call after 5 seconds, then while getting the password, the call gets hangup, due to leg_timeout. To solve this, I've added group_confirm_cancel_timeout=true, but still the other end got hangup immediately, once it reaches the leg_timeout. Here is my dialplan: Here is the commands they I used in NC: connect sendmsg call-command: execute execute-app-name: answer sendmsg call-command:execute execute-app-name: set execute-app-arg: group_confirm_key=exec sendmsg call-command:execute execute-app-name: set execute-app-arg: group_confirm_file=perl /root/bridge.pl sendmsg call-command:execute execute-app-name: set execute-app-arg: group_confirm_cancel_timeout=true sendmsg call-command: execute execute-app-name: bridge execute-app-arg: {group_confirm_cancel_timeout=true}[leg_timeout=10]user/1006 Here is the bridge.pl: #!/usr/bin/perl use freeswitch; use Data::Dumper; our $session; freeswitch::consoleLog("info","Goint to get the digits"); # To simulate the scenario I used sleep here. sleep(30); 1; Here is the FreeSwitch Log. http://pastebin.freeswitch.org/13220 Kindly provide me some inputs. On Fri, Jun 18, 2010 at 7:44 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I said leg timeout beats the group confirm timeouts > > group_confirm_cancel_timeout is a whole different variable, when you set > that to true it will stop all the timeouts as soon as you reach > group_confirm execution > > {group_confirm_cancel_timeout=true}[leg_timeout=10]sofia/foo/foo at bar.com > > > > On Fri, Jun 18, 2010 at 12:50 AM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> Dear Antony, >> >> Also in the leg_timeout wiki >> http://wiki.freeswitch.org/wiki/Variable_leg_timeout, it is stated as >> follows >> >> "If you are using group confirm then you can cancel the timeout by using >> the group_confirm_cancel_timeoutchannel variable." >> >> >> >> On Thu, Jun 17, 2010 at 8:22 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> no there is no way, besides making both timeouts longer. >>> you could file a feature request/bounty to ask for a feature to stop the >>> leg timer when you reach the confirm. >>> >>> >>> On Thu, Jun 17, 2010 at 4:23 AM, Nagalenoj H. wrote: >>> >>>> Anthony, >>>> But, then there is no use. Am I right? Usually, we'll use the >>>> group_confirm_cancel_timeout only when we need to override the leg_timeout. >>>> But it happens in reverse in this case., >>>> >>>> I've tried using the group_confirm_cancel_timeout along with >>>> call_timeout and things happening similar like setting leg_timout. >>>> >>>> Then, tried without setting leg_timeout and call_timeout explicitly. >>>> * In this case if the callee doesn't picks the call, it >>>> disconnects the leg in 30 secs. >>>> * If he answers the call and the script continues to execute, >>>> the leg is disconnected in 60 secs. >>>> >>>> What I need to do is, when the callee picks the call the leg_timeout >>>> should not be accounted more and the leg shouldn't be disconnected because >>>> of leg_timeout after that. >>>> >>>> Any other way of doing this?! >>>> >>>> >>>> >>>> On Tue, Jun 15, 2010 at 10:53 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> leg timeout beats the group confirm timeouts >>>>> >>>>> >>>>> On Tue, Jun 15, 2010 at 12:28 AM, Nagalenoj H. wrote: >>>>> >>>>>> Dear friends, >>>>>> I've tried using the group_confirm_cancel_timeout channel >>>>>> variable. I've written a testing script to get digits before bridging. But, >>>>>> it doesn't seem to be working. >>>>>> >>>>>> My understanding after reading wiki is, >>>>>> * When I dial [leg_timeout=10]user/1005, if he answers before >>>>>> timeout and in the process of giving digits, then the call shouldn't be >>>>>> disconnected after the leg_timeout secs (10 sec in the example). >>>>>> >>>>>> But, When I experiment it, the call is getting disconnected after 10 >>>>>> seconds and it doesn't bother whether the callee has answered the >>>>>> call(Started giving digits) or not answered at all. >>>>>> >>>>>> I've checked it with nc as follows, >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_key=exec >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_file=perl /root/confirm.pl >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_cancel_timeout=1 >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: bridge >>>>>> execute-app-arg: [leg_timeout=10]user/1005 >>>>>> >>>>>> And here is the script, >>>>>> >>>>>> use freeswitch; >>>>>> our $session; >>>>>> my $digit; >>>>>> >>>>>> while(1) { >>>>>> # Wait till response timeout for the first digit. >>>>>> $digit = $session->getDigits(1, "", 10000); >>>>>> freeswitch::consoleLog ("info","Digit>>".$digit."<<"); >>>>>> >>>>>> if (! $session->ready() ) { >>>>>> freeswitch::consoleLog("info","Going to Exit\n"); >>>>>> last; >>>>>> } >>>>>> if (defined $digit and $digit ne "" ) { >>>>>> freeswitch::consoleLog("info","DTMF received: >>>>>> $digit\n"); >>>>>> if ($digit eq '#') { >>>>>> return; >>>>>> } >>>>>> } >>>>>> else { >>>>>> freeswitch::consoleLog("info","Timeout\n"); >>>>>> $session->hangup(); >>>>>> } >>>>>> } >>>>>> 1; >>>>>> >>>>>> If my understanding is right then, I believe there is something wrong >>>>>> with channel_variable. >>>>>> >>>>>> Kindly help me to resolve this. >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> Nagalenoj H. >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> Nagalenoj H. >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > > -- > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100619/0313e01c/attachment-0001.html From msc at freeswitch.org Sat Jun 19 00:37:38 2010 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 19 Jun 2010 00:37:38 -0700 Subject: [Freeswitch-users] Build the mod_unimrcp module In-Reply-To: References: Message-ID: You'll need to visit the web sites for the various vendors. Voxeo has a 30 day trial I think. Most TTS engines that are not horrible will not be free. Sorry... -MC Sent from my iPhone On Jun 18, 2010, at 10:28 PM, "Thangappan.M" wrote: > Where can I get the speech synthesizers? > How do I know the voices which are supported by particular servers? > > > > On Fri, Jun 18, 2010 at 7:38 PM, Christopher Rienzo > wrote: > mod_unimrcp makes FreeSWITCH an MRCP client so it can do TTS/ASR > using an MRCP server. If you do not have (or want to use) an MRCP > server, then this is not the module for you. > > Also, your configuration suggests you are using Nuance Speech > Server, but are requesting a voice, awb, which Nuance doesn't have. > > > On Fri, Jun 18, 2010 at 12:56 AM, Thangappan.M > wrote: > Dear all, > > I am in the process of developing IVR. So just planned to > convert my application to handle the TTS voice engine which is > supported by FreeSWITCH. > Got the mod_unimrcp module which is used to recognize the > speech and synthesize the text to voice. > For building the mod_unimrcp modules done the following > steps. > Uncomment the mod_unimrcp line in the modules.conf > file in the FreeSWITCH source > Given make mod_unimrcp-install command. > In modules.conf.xml uncomment the modue="mod_unimrcp"/> > Configured the following dial plan > > > > > data="tts_engine=unimrcp:nuance5-mrcp1"/> > > --> > > > > > While making the call to 4922 got the following error in the > FreeSWITCH console. > > > [INFO] mod_dialplan_xml.c:418 Processing thangappan->4922 in context > default > > > [NOTICE] mod_dptools.c:717 Channel [sofia/internal/1012 at 192.168.1.222 > ] has been answered > > [INFO] mod_unimrcp.c:1499 speech_handle: name = unimrcp, rate = > 8000, speed = 0, samples = 160, voice = , engine = unimrcp, param = > nuance5-mrcp1 > [INFO] mod_unimrcp.c:1502 voice = awb, rate = 8000 > [NOTICE] mrcp_client.c:549 Create MRCP Handle 0x8b02400 [nuance5- > mrcp1] > > > > [INFO] mrcp_client_session.c:142 Create Channel 0x8b02400 > [INFO] mrcp_client_session.c:398 Receive App Request 0x8b02400 > [2] > [NOTICE] rtsp_client.c:255 Create RTSP Handle 0x8b04408 > > > [INFO] mrcp_client.c:901 Add MRCP Handle 0x8b02400 > > [NOTICE] mrcp_client_session.c:718 Add Control Channel 0x8b02400 > > [INFO] mrcp_client_session.c:420 Send Offer 0x8b02400 [c:0 a: > 1 v:0] > [ERR] mod_unimrcp.c:965 (TTS-0) Timed out waiting for channel to > be ready > > > > [ERR] switch_ivr_play_say.c:2104 Invalid TTS module! > [NOTICE] switch_core_state_machine.c:185 sofia/internal/1012 at 192.168.1.222 > has executed the last dialplan instruction, hanging up. > > > > [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/ > 1012 at 192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING] > [NOTICE] switch_core_session.c:1179 Session 1 (sofia/internal/1012 at 192.168.1.222 > ) Ended > > > > [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/ > 1012 at 192.168.1.222 [CS_DESTROY] > [NOTICE] switch_channel.c:669 New Channel sofia/internal/1012 at 192.168.1.222 > [49ed5afa-79f0-11df-b531-3553f3a65c3c] > > > > So need to find a solution for that. > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.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 > > > > > -- > Regards, > Thangappan.M > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100619/517655a4/attachment.html From roland at haenel.me Sat Jun 19 02:34:06 2010 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Sat, 19 Jun 2010 11:34:06 +0200 Subject: [Freeswitch-users] Matching CHANNEL_CREATE event to previous originate command Message-ID: Hello everyone, I'm using mod_event_socket in inbound mode to 'remote control' a FreeSwitch instance. I'm using the raw TCP interface (not any of the available language bindings). My problem now is, if I do a command like bgapi originate sofia/gateway/mygw/0123456789 &park and try to match the channel which is created by this command to the command itself, this seems to be very difficult. I cannot do it on timing only, because many channels may be created in a very short time and the order of originate-CHANNEL_CREATE will probably not be consistent. Is there any 'best practice' how to solve this problem? One solution seems to be to set {origination_uuid=x-y-z-k} in the originate section, but this does not reliably work with failover routing (for example, if I specify two gateways and the call gets rejected on the first and then originated on the second, the origination_uuid will not be the same any more). Greetings, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100619/c530fc85/attachment.html From mcampbellsmith at gmail.com Sat Jun 19 07:39:17 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 20 Jun 2010 00:39:17 +1000 Subject: [Freeswitch-users] G729 Encoder Error Message-ID: Hi! I just installed the G729 licensed codec and when I make a call to FS, I get an Encoder error: 2010-06-20 00:35:02.623715 [DEBUG] switch_ivr_originate.c:1159 Play Ringback Tone [v=-7;%(400,200,413,438);%(400,2000,413,438)] 2010-06-20 00:35:02.678615 [INFO] mod_com_g729.c:117 ENCODER CREATE - 0x86f1fb0 0xb7032fc0 I don't hear the ringback tone. Later after hanging up the call, I see more encoder errors: 2010-06-20 00:35:08.972764 [NOTICE] switch_core_session.c:1182 Session 3 (sofia/external/xx at 201.84.242.81) Ended 2010-06-20 00:35:08.976339 [NOTICE] switch_core_session.c:1184 Close Channel sofia/external/xx at 201.84.242.81 [CS_DESTROY] 2010-06-20 00:35:08.989462 [DEBUG] switch_core_state_machine.c:428 (sofia/external/xx at 201.84.242.81) Running State Change CS_DESTROY 2010-06-20 00:35:08.995797 [DEBUG] switch_core_state_machine.c:439 (sofia/external/xx at 201.84.242.81) State DESTROY 2010-06-20 00:35:08.999238 [DEBUG] mod_sofia.c:341 sofia/external/xx at 201.84.242.81 SOFIA DESTROY 2010-06-20 00:35:09.004366 [INFO] mod_com_g729.c:76 ENCODER DESTROYX - 0x86f1f58 (nil) 2010-06-20 00:35:09.005537 [INFO] mod_com_g729.c:77 DECODER DESTROYX - 0x86f1f58 (nil) 2010-06-20 00:35:09.007634 [INFO] mod_com_g729.c:76 ENCODER DESTROYX - 0x86f1fb0 0xb7032fc0 2010-06-20 00:35:09.012505 [INFO] mod_com_g729.c:77 DECODER DESTROYX - 0x86f1fb0 (nil) 2010-06-20 00:35:09.032363 [INFO] mod_com_g729.c:81 ENCODER DESTROY - 0x86f1fb0 0xb7032fc FS version is FreeSWITCH Version 1.0.head (git-8ad17db 2010-04-09 00:43:24 -0300). I'm updating as we speak to see if this solves the issue. Could that be the issue? Thanks From rupa at rupa.com Sat Jun 19 09:15:41 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Sat, 19 Jun 2010 11:15:41 -0500 Subject: [Freeswitch-users] mod_limit / db / hash changes - If you update action is required Message-ID: If you've been paying attention to the conf call or irc you'll know this change was coming. I've committed a significant change to how limits are managed in FreeSWITCH. This change is NOT backwards compatible. It requires one to: 0) git pull (of course) 1) rerun ./configure to generate new Makefiles for new modules 2) edit modules.conf, remove mod_limit and add mod_db and mod_hash 3) modify conf/autoload_configs/modules.conf.xml, remove mod_limit and add mod_db and mod_hash 4) create/edit conf/autoload_configs/db.conf.xml if using ODBC In your dialplan, anywhere you are using limit, you need to add db to the data line. Anywhere you are using limit_hash you need to remove _hash from application and add hash to the front of the data line. Same pattern applies to any API usage of limit you may be using. Read the wiki at: http://wiki.freeswitch.org/wiki/Limit for documentation on limit. Ok, why the change? limit has been moved into core with support for pluggable backend limit providers. This allows one to develop limit backends relatively easily and provides a more structured interface to these different limit backends. Expect to see backends provided by more than just db and hash. It also always bothered me that db and hash were part of the mod_limit module. They are not standalone and provide functionality specific to themselves. I've updated the wiki with documentation. I'll continue to update the wiki as necessary, feel free to do so yourself if there is something unclear or missing. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100619/16dfb9a1/attachment.html From anthony.minessale at gmail.com Sat Jun 19 09:41:41 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Jun 2010 11:41:41 -0500 Subject: [Freeswitch-users] G729 Encoder Error In-Reply-To: References: Message-ID: Did you uninstall the regular mod_g729 that is passthru only before loading the commerical one? On Jun 19, 2010 9:46 AM, "Mark Campbell-Smith" wrote: Hi! I just installed the G729 licensed codec and when I make a call to FS, I get an Encoder error: 2010-06-20 00:35:02.623715 [DEBUG] switch_ivr_originate.c:1159 Play Ringback Tone [v=-7;%(400,200,413,438);%(400,2000,413,438)] 2010-06-20 00:35:02.678615 [INFO] mod_com_g729.c:117 ENCODER CREATE - 0x86f1fb0 0xb7032fc0 I don't hear the ringback tone. Later after hanging up the call, I see more encoder errors: 2010-06-20 00:35:08.972764 [NOTICE] switch_core_session.c:1182 Session 3 (sofia/external/xx at 201.84.242.81) Ended 2010-06-20 00:35:08.976339 [NOTICE] switch_core_session.c:1184 Close Channel sofia/external/xx at 201.84.242.81 [CS_DESTROY] 2010-06-20 00:35:08.989462 [DEBUG] switch_core_state_machine.c:428 (sofia/external/xx at 201.84.242.81) Running State Change CS_DESTROY 2010-06-20 00:35:08.995797 [DEBUG] switch_core_state_machine.c:439 (sofia/external/xx at 201.84.242.81) State DESTROY 2010-06-20 00:35:08.999238 [DEBUG] mod_sofia.c:341 sofia/external/xx at 201.84.242.81 SOFIA DESTROY 2010-06-20 00:35:09.004366 [INFO] mod_com_g729.c:76 ENCODER DESTROYX - 0x86f1f58 (nil) 2010-06-20 00:35:09.005537 [INFO] mod_com_g729.c:77 DECODER DESTROYX - 0x86f1f58 (nil) 2010-06-20 00:35:09.007634 [INFO] mod_com_g729.c:76 ENCODER DESTROYX - 0x86f1fb0 0xb7032fc0 2010-06-20 00:35:09.012505 [INFO] mod_com_g729.c:77 DECODER DESTROYX - 0x86f1fb0 (nil) 2010-06-20 00:35:09.032363 [INFO] mod_com_g729.c:81 ENCODER DESTROY - 0x86f1fb0 0xb7032fc FS version is FreeSWITCH Version 1.0.head (git-8ad17db 2010-04-09 00:43:24 -0300). I'm updating as we speak to see if this solves the issue. Could that be the issue? Thanks _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100619/65fb23eb/attachment-0001.html From tayeb.meftah at gmail.com Sun Jun 20 08:45:09 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sun, 20 Jun 2010 17:45:09 +0200 Subject: [Freeswitch-users] question about NibbleBill In-Reply-To: <61FDF212ED8B4D42AE335E4211226EFE@MOBILEE1705> References: <61FDF212ED8B4D42AE335E4211226EFE@MOBILEE1705> Message-ID: <4C1E3785.7010301@gmail.com> hi, use a lua or a javascript script to lookup your user and set the rate acording to the account Le 19/06/2010 01:07, Madovsky a ?crit : > hi there, > I'm trying to create user who can call for free and other not. > I set nobal_action in nibblebill conf to an extension in my default > dialplan > but if the user is at $0 for special case I would authorize the call > for free. > is it possible ? > Thanks > F > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100620/7e3d5ce9/attachment.html From freeswitch-users at skelly.ca Fri Jun 18 18:58:26 2010 From: freeswitch-users at skelly.ca (FreeSWITCH-Users) Date: Fri, 18 Jun 2010 18:58:26 -0700 Subject: [Freeswitch-users] Mobile callback + DISA Message-ID: Hi all, I am new to FS but have been working with * for a while.. I have * setup for callback+DISA used by family for cellular minutes. Have been looking at FS this past month or so as I'd like to setup FS on my router with DD-WRT + FS. Presently I have figured out how to setup disa.js for DISA on my cell BUT I can't seem to find anything out on the web about configuring callback. Has anyone figured out how to setup callback+DISA? Any help is appreciated, links or any ideas would help me over this hurdle. Cheers, S. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100618/9b843526/attachment.html From xengelpublicx at gmail.com Sat Jun 19 15:42:44 2010 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Sun, 20 Jun 2010 02:42:44 +0400 Subject: [Freeswitch-users] continue_on_fail and hangup_after_bridge with transfer Message-ID: I'm trying to make a dialplan: The logic of his work: if unavailable ilti gateway to the next. if not available all the gateway to go to the extension transfer. A problem arises when user_busy on the phone: ------------------------------------------------------------------------ SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 192.168.50.11:5080;rport;branch=z9hG4bK1X4r00t93KmFg From: "Vladimir Elizarov" ;tag=yj15KpSQvK6eF To: ;tag=2a5549004a431d10ff00001a646aff45 Call-ID: 1317949e-f694-122d-8fb6-00163efcbed2 CSeq: 132374757 INVITE Server: MERA MSIP v.1.0.2 Reason: Q.850;cause=17;text="User busy" Content-Length: 0 instead of making hangup, dialplan is going to transfer. Constant despite continue_on_fail = NORMAL_TEMPORARY_FAILURE, TIMEOUT, NO_ROUTE_DESTINATION" What's the problem? freeswitch 1.0.6 (git 19 june 2010) -- Best regards, Vladimir Elizarov From rupa at rupa.com Sat Jun 19 18:38:34 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Sat, 19 Jun 2010 20:38:34 -0500 Subject: [Freeswitch-users] mod_lcr update - read if you use custom_sql Message-ID: mod_lcr has been updated. commit log: mod_lcr update. MODAPP-340, MODAPP-355 arbitrary b-leg vars, limit support, lcr/ endpoint custom_sql now is field based rather than position based NOTE: custom_sql is incompat with prior version arbitrary b-leg vars enable integration with mod_nibblebill. Wiki has been updated: http://wiki.freeswitch.org/wiki/Mod_lcr This change is backwards compatible EXCEPT if you use custom_sql. If you use custom_sql you must update your sql to use field aliases (SQL : AS). The module no longer uses position in the custom sql for fields. So, if you use custom_sql, reread: http://wiki.freeswitch.org/wiki/Mod_lcr#Custom_SQL Ok, I'm done breaking things for the day. :) -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100619/f2557064/attachment.html From infos at madovsky.org Sat Jun 19 19:44:42 2010 From: infos at madovsky.org (Madovsky) Date: Sat, 19 Jun 2010 22:44:42 -0400 Subject: [Freeswitch-users] mod_lcr update - read if you use custom_sql References: Message-ID: <5417357135F44CC1838BFDE07AFA433D@MOBILEE1705> ok hanks Rupa ----- Original Message ----- From: Rupa Schomaker To: freeswitch-users ; freeswitch-dev Sent: Saturday, June 19, 2010 9:38 PM Subject: [Freeswitch-users] mod_lcr update - read if you use custom_sql mod_lcr has been updated. commit log: mod_lcr update. MODAPP-340, MODAPP-355 arbitrary b-leg vars, limit support, lcr/ endpoint custom_sql now is field based rather than position based NOTE: custom_sql is incompat with prior version arbitrary b-leg vars enable integration with mod_nibblebill. Wiki has been updated: http://wiki.freeswitch.org/wiki/Mod_lcr This change is backwards compatible EXCEPT if you use custom_sql. If you use custom_sql you must update your sql to use field aliases (SQL : AS). The module no longer uses position in the custom sql for fields. So, if you use custom_sql, reread: http://wiki.freeswitch.org/wiki/Mod_lcr#Custom_SQL Ok, I'm done breaking things for the day. :) -- -Rupa ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100619/fbab5674/attachment.html From babak.freeswitch at gmail.com Sat Jun 19 21:29:51 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Sun, 20 Jun 2010 08:59:51 +0430 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: References: Message-ID: thanx for ur answers and suggestions. I've simulated the cisco call manager in a an application and somehow implemented the extension mobility (thanx to freeswitch strong scripting modules :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100620/b664ec18/attachment.html From mike at jerris.com Sat Jun 19 23:52:44 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 20 Jun 2010 02:52:44 -0400 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: References: Message-ID: <1150252E-73A1-4C6C-B330-8E0A460FF2A2@jerris.com> Are you sharing this? On Jun 20, 2010, at 12:29 AM, babak yakhchali wrote: > thanx for ur answers and suggestions. I've simulated the cisco call manager in a an application and somehow implemented the extension mobility (thanx to freeswitch strong scripting modules :) From david.ponzone at gmail.com Sun Jun 20 01:13:55 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Sun, 20 Jun 2010 10:13:55 +0200 Subject: [Freeswitch-users] Mobile callback + DISA In-Reply-To: References: Message-ID: You probably won't find anything existing to do this, but this can be easily implemented in a script. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 19/06/2010 ? 03:58, FreeSWITCH-Users a ?crit : > Hi all, > > I am new to FS but have been working with * for a while.. I have * > setup for callback+DISA used by family for cellular minutes. Have > been looking at FS this past month or so as I'd like to setup FS on > my router with DD-WRT + FS. Presently I have figured out how to > setup disa.js for DISA on my cell BUT I can't seem to find anything > out on the web about configuring callback. Has anyone figured out > how to setup callback+DISA? > > Any help is appreciated, links or any ideas would help me over this > hurdle. > > Cheers, > > S. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100620/f610fe18/attachment-0001.html From stephen at mymessage.us Sun Jun 20 02:42:22 2010 From: stephen at mymessage.us (Stephen Cattaneo) Date: Sun, 20 Jun 2010 05:42:22 -0400 Subject: [Freeswitch-users] Mobile callback + DISA In-Reply-To: References: Message-ID: This is just to get you in a direction. After if (dtmf.digits == pin || pin.length == 0) in disa.js you would do something like. hangup wait x seconds call you back, probably based on session.caller_id_number by creating a new session. then finish the rest of the disa code. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print On Fri, Jun 18, 2010 at 9:58 PM, FreeSWITCH-Users < freeswitch-users at skelly.ca> wrote: > Hi all, > > I am new to FS but have been working with * for a while.. I have * setup > for callback+DISA used by family for cellular minutes. Have been looking at > FS this past month or so as I'd like to setup FS on my router with DD-WRT + > FS. Presently I have figured out how to setup disa.js for DISA on my cell > BUT I can't seem to find anything out on the web about configuring callback. > Has anyone figured out how to setup callback+DISA? > > Any help is appreciated, links or any ideas would help me over this hurdle. > > Cheers, > > S. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100620/7a298d67/attachment.html From david.ponzone at gmail.com Sun Jun 20 03:08:29 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Sun, 20 Jun 2010 12:08:29 +0200 Subject: [Freeswitch-users] Mobile callback + DISA In-Reply-To: References: Message-ID: <08B0CBA1-7D27-4E81-A7C3-01A784F94B93@gmail.com> And you can probably setup a callback to Caller-ID without answering the call too, for known Caller-IDs. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 20/06/2010 ? 11:42, Stephen Cattaneo a ?crit : > This is just to get you in a direction. After if (dtmf.digits == pin > || pin.length == 0) in disa.js you would do something like. > > hangup > wait x seconds > call you back, probably based on session.caller_id_number by > creating a new session. > then finish the rest of the disa code. > > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print > > > > On Fri, Jun 18, 2010 at 9:58 PM, FreeSWITCH-Users > wrote: > Hi all, > > I am new to FS but have been working with * for a while.. I have * > setup for callback+DISA used by family for cellular minutes. Have > been looking at FS this past month or so as I'd like to setup FS on > my router with DD-WRT + FS. Presently I have figured out how to > setup disa.js for DISA on my cell BUT I can't seem to find anything > out on the web about configuring callback. Has anyone figured out > how to setup callback+DISA? > > Any help is appreciated, links or any ideas would help me over this > hurdle. > > Cheers, > > S. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100620/91c04f3a/attachment.html From petedao at gmail.com Sun Jun 20 03:57:14 2010 From: petedao at gmail.com (Pete Kay) Date: Sun, 20 Jun 2010 18:57:14 +0800 Subject: [Freeswitch-users] check if transcoding is needed Message-ID: Hi, Is there anyway to detect/check if codec transcoding is needed? What I would like to do is to set "bypass_media=true" if no transcoding is needed. Any suggestion on how / when should bypass_media be turned on/off? thanks, P From mcampbellsmith at gmail.com Sun Jun 20 04:45:35 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 20 Jun 2010 21:45:35 +1000 Subject: [Freeswitch-users] G729 Encoder Error In-Reply-To: References: Message-ID: Thanks Anthony. I actually had followed the instructions in the txt file, and had unloaded mod g729. Anyway, a restart of FS solved the problem (and its up to the latest git checkout) Cheers On Sun, Jun 20, 2010 at 2:41 AM, Anthony Minessale wrote: > Did you uninstall the regular mod_g729 that is passthru only before loading > the commerical one? > > On Jun 19, 2010 9:46 AM, "Mark Campbell-Smith" > wrote: > > Hi! > > I just installed the G729 licensed codec and when I make a call to FS, > I get an Encoder error: > > 2010-06-20 00:35:02.623715 [DEBUG] switch_ivr_originate.c:1159 Play > Ringback Tone [v=-7;%(400,200,413,438);%(400,2000,413,438)] > 2010-06-20 00:35:02.678615 [INFO] mod_com_g729.c:117 ENCODER CREATE - > 0x86f1fb0 0xb7032fc0 > > I don't hear the ringback tone. > > Later after hanging up the call, I see more encoder errors: > > 2010-06-20 00:35:08.972764 [NOTICE] switch_core_session.c:1182 Session > 3 (sofia/external/xx at 201.84.242.81) Ended > 2010-06-20 00:35:08.976339 [NOTICE] switch_core_session.c:1184 Close > Channel sofia/external/xx at 201.84.242.81 [CS_DESTROY] > 2010-06-20 00:35:08.989462 [DEBUG] switch_core_state_machine.c:428 > (sofia/external/xx at 201.84.242.81) Running State Change CS_DESTROY > 2010-06-20 00:35:08.995797 [DEBUG] switch_core_state_machine.c:439 > (sofia/external/xx at 201.84.242.81) State DESTROY > 2010-06-20 00:35:08.999238 [DEBUG] mod_sofia.c:341 > sofia/external/xx at 201.84.242.81 SOFIA DESTROY > 2010-06-20 00:35:09.004366 [INFO] mod_com_g729.c:76 ENCODER DESTROYX - > 0x86f1f58 (nil) > 2010-06-20 00:35:09.005537 [INFO] mod_com_g729.c:77 DECODER DESTROYX - > 0x86f1f58 (nil) > 2010-06-20 00:35:09.007634 [INFO] mod_com_g729.c:76 ENCODER DESTROYX - > 0x86f1fb0 0xb7032fc0 > 2010-06-20 00:35:09.012505 [INFO] mod_com_g729.c:77 DECODER DESTROYX - > 0x86f1fb0 (nil) > 2010-06-20 00:35:09.032363 [INFO] mod_com_g729.c:81 ENCODER DESTROY - > 0x86f1fb0 0xb7032fc > > FS version is FreeSWITCH Version 1.0.head (git-8ad17db 2010-04-09 > 00:43:24 -0300). ?I'm updating as we speak to see if this solves the > issue. ?Could that be the issue? > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > From infos at madovsky.org Sun Jun 20 09:38:59 2010 From: infos at madovsky.org (Madovsky) Date: Sun, 20 Jun 2010 12:38:59 -0400 Subject: [Freeswitch-users] check if transcoding is needed References: Message-ID: <4038C46B1BF343ABAC9E5EDF853695FB@MOBILEE1705> see with disable_transcoding on wiki bypass_media is another thing ----- Original Message ----- From: "Pete Kay" To: Sent: Sunday, June 20, 2010 6:57 AM Subject: [Freeswitch-users] check if transcoding is needed > Hi, > > Is there anyway to detect/check if codec transcoding is needed? What > I would like to do is to set "bypass_media=true" if no transcoding is > needed. > > Any suggestion on how / when should bypass_media be turned on/off? > > thanks, > P > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 mranga at gmail.com Sun Jun 20 12:47:03 2010 From: mranga at gmail.com (M. Ranganathan) Date: Sun, 20 Jun 2010 15:47:03 -0400 Subject: [Freeswitch-users] Start/stop recording controls and multiple recordings of a single conference. Message-ID: Hello, I have a couple of questions about the behavior of conference recording: 1. In Looking at the semantics of norecord as stated in, http://wiki.freeswitch.org/wiki/Mod_conference it is not clear to me whether or not it is possible to start/stop/restart recording of an ongoing conference. Any clarifications would be appreciated. 2. Can you have multiple recordings for a given conference? i.e. conference foo record /tmp/recording1.wav and conference foo record /tmp/recording2.wav at the same time for a single ongoing conference. Thank you in advance for any help. Regards, Ranga -- M. Ranganathan From rupa at rupa.com Sun Jun 20 17:08:06 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Sun, 20 Jun 2010 19:08:06 -0500 Subject: [Freeswitch-users] mod_limit / db / hash changes - If you update action is required In-Reply-To: References: Message-ID: After some prodding I've checked in a mod_limit which enables a backwards compatible mode for limit behavior. It is a bit noisy with warnings about deprecated calls -- this is to encourage you to migrate. It will auto-load mod_db and mod_hash and supports the older calling convention (limit with not backend specified, limit_hash, etc). You till need to edit modules.conf to enable building mod_db and mod_hash but your runtime config doesn't need to be updated until you are ready to do so. unloading mod_limit will turn off the older behavior. On Sat, Jun 19, 2010 at 11:15 AM, Rupa Schomaker wrote: > If you've been paying attention to the conf call or irc you'll know this > change was coming. I've committed a significant change to how limits are > managed in FreeSWITCH. This change is NOT backwards compatible. It > requires one to: > > 0) git pull (of course) > 1) rerun ./configure to generate new Makefiles for new modules > 2) edit modules.conf, remove mod_limit and add mod_db and mod_hash > 3) modify conf/autoload_configs/modules.conf.xml, remove mod_limit and add > mod_db and mod_hash > 4) create/edit conf/autoload_configs/db.conf.xml if using ODBC > > In your dialplan, anywhere you are using limit, you need to add db to the > data line. Anywhere you are using limit_hash you need to remove _hash from > application and add hash to the front of the data line. Same pattern > applies to any API usage of limit you may be using. > > Read the wiki at: http://wiki.freeswitch.org/wiki/Limit for documentation > on limit. > > Ok, why the change? > > limit has been moved into core with support for pluggable backend limit > providers. This allows one to develop limit backends relatively easily and > provides a more structured interface to these different limit backends. > Expect to see backends provided by more than just db and hash. > > It also always bothered me that db and hash were part of the mod_limit > module. They are not standalone and provide functionality specific to > themselves. > > I've updated the wiki with documentation. I'll continue to update the wiki > as necessary, feel free to do so yourself if there is something unclear or > missing. > > -- > -Rupa > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100620/b608433a/attachment-0001.html From abu.4000 at gmail.com Mon Jun 21 03:33:45 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Mon, 21 Jun 2010 16:03:45 +0530 Subject: [Freeswitch-users] monitor the chat Message-ID: Dear all , My requirement is to monitor the chatting of all the extensions using outbound or inbound extensions and I need to log those informations. I don't know how to do this using outbound , In inbound I could not get any events when the chatting happens. Please let me know how to do this. Thanks in advance ! -- BEST REGARDS N.ABUBACKER SOFTWARE ENGINEER BK SYSTEMS (P) LTD CHENNAI-23 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/c95d4928/attachment.html From babak.freeswitch at gmail.com Mon Jun 21 03:54:39 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Mon, 21 Jun 2010 15:24:39 +0430 Subject: [Freeswitch-users] extension mobility using freeswitch In-Reply-To: <1150252E-73A1-4C6C-B330-8E0A460FF2A2@jerris.com> References: <1150252E-73A1-4C6C-B330-8E0A460FF2A2@jerris.com> Message-ID: yes as soon as I'm sure it's working I will share it :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/4f52daab/attachment.html From brian at freeswitch.org Mon Jun 21 06:46:51 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Jun 2010 08:46:51 -0500 Subject: [Freeswitch-users] G729 Encoder Error In-Reply-To: References: Message-ID: <8DDED232-4E7A-41D0-A8EF-ABB15C755F7D@freeswitch.org> Are you running under Xen? /b On Jun 20, 2010, at 6:45 AM, Mark Campbell-Smith wrote: > Thanks Anthony. I actually had followed the instructions in the txt > file, and had unloaded mod g729. > > Anyway, a restart of FS solved the problem (and its up to the latest > git checkout) > > Cheers From neilp at cs.stanford.edu Mon Jun 21 06:51:59 2010 From: neilp at cs.stanford.edu (Neil Patel) Date: Mon, 21 Jun 2010 19:21:59 +0530 Subject: [Freeswitch-users] span not defined error In-Reply-To: References: Message-ID: Hi Moises, I recently added a second PRI line to my A102 and followed your instructions below (modified /etc/wanpipe/smg_pri.conf; openzap.conf as shown below) and can successfully dial out with g1, but not the recently added second line set on g2. Below is the error we get: freeswitch at otalo> originate openzap/smg_prid/a/@g2 &echo 2010-06-21 19:15:59.591930 [WARNING] ozmod_sangoma_boost.c:344 TX EVENT: CALL_START:(80) [w1g1] CSid=3 Seq=4 Cn=[N/A] Cd=[9586550654] Ci=[N/A] Rdnis=[] 2010-06-21 19:15:59.624992 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT (N): CALL_START_ACK:(81) [w2g1] Rc=0 CSid=3 Seq=6 2010-06-21 19:15:59.625987 [NOTICE] switch_channel.c:675 New Channel OpenZAP/1:31/@g2 [19ae26fb-f9e4-480a-9cc7-11fa91b8bb1c] 2010-06-21 19:15:59.724996 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT (N): CALL_STOPPED:(85) [w2g1] Rc=3 CSid=3 Seq=7 2010-06-21 19:15:59.724996 [NOTICE] mod_openzap.c:1935 Hangup OpenZAP/1:31/@g2 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] -ERR NO_ROUTE_DESTINATION Seems like a config issue, but not sure what I'm missing. Any ideas? Thanks in advance, Neil On Sun, May 16, 2010 at 10:50 AM, Moises Silva wrote: > Hi Again Neil, > > I just noticed your dial string is incorrect. The correct syntax is: > > OpenZAP///[number] > > The span and chan code are mandatory. The number is optional ( FXS channels > do not require a number, they just ring the FXO device connected to them). > > The span is either a number ( span id, the id is a number assigned in the > order in which the span is defined in openzap.conf ) or a span name also as > specified in the [span wanpipe ] line in openzap.conf > > The chan code is either a number ( for spans that support individual > channel selection, boost is NOT one of them ), or a channel hunting mode, > there is currently 2 modes, "a" is top down and "A" is bottom up. > > So, this is a valid string for you case: > > OpenZAP/smg_prid/a/ > > In the specific case of boost in socket mode ( openzap only supports socket > mode ) the number may contain @gX where X is a group ( for hunting as > configured in /etc/wanpipe/smg_pri.conf). Boost signaling are a special case > because the hunting for channels is not done within FreeSWITCH but in > sangoma_prid binary ( in the new OpenZAP version called FreeTDM this has > changed depending on configuration). > > Bottom line, this should work: > > OpenZAP/smg_prid/a/1234 at g1 > > If you have g1 configured in /etc/wanpipe/smg_pri.conf > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > > On Sat, May 15, 2010 at 6:26 PM, Neil Patel wrote: > >> Span was originally in the boost section when I got this error, so I >> thought I'd try it in analog and both. None work. >> >> -Neil >> >> On Sat, May 15, 2010 at 3:18 PM, Moises Silva wrote: >> >>> Why do you have 2 spans in openzap.conf.xml with the same name, in both >>> the boost and analog sections? >>> Moises Silva >>> Senior Software Engineer >>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R >>> 9T3 Canada >>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com >>> >>> >>> On Sat, May 15, 2010 at 3:52 PM, Neil Patel wrote: >>> >>>> Hi All, >>>> >>>> I am trying to dial out over my PRI, and am getting this error: >>>> >>>> 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED! >>>> 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels >>>> available >>>> 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot >>>> create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION] >>>> >>>> >>>> This is my openzap.conf: >>>> >>>> [span wanpipe smg_prid] >>>> name => smg_prid >>>> trunk_type =>e1 >>>> b-channel => 1:1-15 >>>> b-channel => 1:17-31 >>>> trunk_type =>e1 >>>> b-channel => 2:1-15 >>>> b-channel => 2:17-31 >>>> >>>> >>>> This is my openzap.conf.xml: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> And here is the lua code I'm using to dial out: >>>> >>>> sessiondata = "OpenZAP/smg_prid/" >>>> new_session = freeswitch.Session(sessiondata) >>>> >>>> >>>> What am I missing here? >>>> Thanks, >>>> Neil >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100621/3599ca5c/attachment.html From mranga at gmail.com Mon Jun 21 07:17:07 2010 From: mranga at gmail.com (M. Ranganathan) Date: Mon, 21 Jun 2010 10:17:07 -0400 Subject: [Freeswitch-users] Second session with an endpoint. Message-ID: Hello, I want to mute one RTP session with a given endpoint and establish a second session with that endpoint when the first is muted. Will FreeSWITCH allow me to do this? Is a Muted session still considered to be "active". Thanks Ranga -- M. Ranganathan From christian.loeschenkohl at xpirio.com Mon Jun 21 08:30:53 2010 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 21 Jun 2010 17:30:53 +0200 Subject: [Freeswitch-users] t.38 in newer fs versions - t38-passthru Message-ID: <4C1F85AD.1000902@xpirio.com> hello has anybody tried the config parameter t38-passthru in a sip profile? our fs server core dumped today after i did set the option an did a reladxml + a profile rescan does this option bring back the old behavior that the proxy mode provided? the actual behavior (actual git from 17.06.2010) is not useable. t.38 isn't passed through without modification and fax sending fails almost every time. in e.g. version 1.0.6 t.38 in proxy mode worked 100% reliable br -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From stephen at mymessage.us Mon Jun 21 08:33:49 2010 From: stephen at mymessage.us (Stephen Cattaneo) Date: Mon, 21 Jun 2010 11:33:49 -0400 Subject: [Freeswitch-users] session variable question Message-ID: Are the following the same thing? i just want to make sure because i've seen both in example scripts. session.setVariable("hangup_after_bridge",true); session.execute("set","hangup_after_bridge=true"); Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/33427f4a/attachment.html From brian at freeswitch.org Mon Jun 21 08:39:17 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Jun 2010 10:39:17 -0500 Subject: [Freeswitch-users] t.38 in newer fs versions - t38-passthru In-Reply-To: <4C1F85AD.1000902@xpirio.com> References: <4C1F85AD.1000902@xpirio.com> Message-ID: Did you follow protocol on how to collect a backtrace and report it on jira? http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Jun 21, 2010, at 10:30 AM, Christian L?schenkohl wrote: > hello > > has anybody tried the config parameter t38-passthru in a sip profile? > our fs server core dumped today after i did set the option an did a reladxml + a profile rescan > > does this option bring back the old behavior that the proxy mode provided? > the actual behavior (actual git from 17.06.2010) is not useable. t.38 isn't passed through > without modification and fax sending fails almost every time. > in e.g. version 1.0.6 t.38 in proxy mode worked 100% reliable > > br > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP From brian at freeswitch.org Mon Jun 21 08:40:13 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Jun 2010 10:40:13 -0500 Subject: [Freeswitch-users] session variable question In-Reply-To: References: Message-ID: <5E6A6053-C10F-4CC2-9497-FB78CD7F888D@freeswitch.org> Yes they are the same. /b -Save Bandwidth, Highlight before you click reply. On Jun 21, 2010, at 10:33 AM, Stephen Cattaneo wrote: > Are the following the same thing? i just want to make sure because i've seen both in example scripts. > > session.setVariable("hangup_after_bridge",true); > session.execute("set","hangup_after_bridge=true"); From Prometheus001 at gmx.net Mon Jun 21 08:49:53 2010 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 21 Jun 2010 17:49:53 +0200 Subject: [Freeswitch-users] how to get rid of second P-Asserted-Identity? Message-ID: <4C1F8A21.7030702@gmx.net> When receiving a call from PSTN and forwarding it to another PSTN number and setting P-Asserted-Identity header, I found that 2 P-Asserted-Identity headers are present in the INVITE message. As the target provider only checks the first one, the call is denied. The first P-Asserted-Identity header is from the the incoming call. The second is set in our dialplan. Is there achance to drop the first header part? Best regards Peter Here's the dialplan: ]]> ]]> Here is the corresponding INVITE. INVITE sip:0162xxxxxxxxxxxxxx at sip1.my.domain.de SIP/2.0. Via: SIP/2.0/UDP 82.xxx.xx.1x3:5080;rport;branch=z9hG4bKjD7UvXB8XF5jD. Max-Forwards: 29. From: "02x139xxxxx" ;tag=U4v0ypBB1X78F. To: . Call-ID: 1cdc72b9-f7ea-122d-e685-001ec9b9dd2d. CSeq: 132448209 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15434M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 320. X-FS-Support: update_display. P-Asserted-Identity: "02x139xxxxx" . (from incoing call) P-Asserted-Identity: . (set in dialplan) P-Preferred-Identity: . . v=0. o=FreeSWITCH 1277106526 1277106527 IN IP4 82.xxx.xx.1x3. s=FreeSWITCH. c=IN IP4 82.xxx.xx.1x3. t=0 0. m=audio 26564 RTP/AVP 8 0 98 3 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:98 SPEEX/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. From infos at madovsky.org Mon Jun 21 09:47:21 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 21 Jun 2010 12:47:21 -0400 Subject: [Freeswitch-users] voicemail wav filename Message-ID: <1373EEB45A00408FBC979E1F027A71A8@MOBILEE1705> Hi, where can I customize the wav filename sent from voicemail ? Thanks F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/f6aa81d2/attachment.html From toqeer83 at gmail.com Mon Jun 21 09:56:40 2010 From: toqeer83 at gmail.com (toqeer ali) Date: Mon, 21 Jun 2010 21:56:40 +0500 Subject: [Freeswitch-users] Freeswitch and freepbxv3 Message-ID: Hi all, I have a confusion... People are talking about freepbxv3 and freeswitch. But i do not understand what it is.Whether it is an graphical interface for freeswitch? Please explain if someone understand this, Thanks -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/12a734a5/attachment.html From infos at madovsky.org Mon Jun 21 10:02:46 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 21 Jun 2010 13:02:46 -0400 Subject: [Freeswitch-users] Freeswitch and freepbxv3 References: Message-ID: <9EEE8101DEFE40D08224848E925E7E46@MOBILEE1705> yes ----- Original Message ----- From: toqeer ali To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 21, 2010 12:56 PM Subject: [Freeswitch-users] Freeswitch and freepbxv3 Hi all, I have a confusion... People are talking about freepbxv3 and freeswitch. But i do not understand what it is.Whether it is an graphical interface for freeswitch? Please explain if someone understand this, Thanks -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/155ab316/attachment.html From gmaruzz at celliax.org Mon Jun 21 10:04:36 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 21 Jun 2010 19:04:36 +0200 Subject: [Freeswitch-users] Freeswitch and freepbxv3 In-Reply-To: References: Message-ID: yes it is http://www.freepbx.org/v3/roadmap On Mon, Jun 21, 2010 at 6:56 PM, toqeer ali wrote: > > Hi all, > > I have a confusion... > > People are talking about freepbxv3 and freeswitch. But i do not understand > what it is.Whether it is an graphical interface for freeswitch? > > Please explain if someone understand this, > > Thanks > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: ? ? +92 321 9059916 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From moises.silva at gmail.com Mon Jun 21 10:09:42 2010 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 21 Jun 2010 13:09:42 -0400 Subject: [Freeswitch-users] span not defined error In-Reply-To: References: Message-ID: Check the sangoma_prid log, from FS point of view there is nothing wrong. Make sure you have defined g2 properly. The sangoma_prid log is /var/log/sangoma_mgd.log Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Mon, Jun 21, 2010 at 9:51 AM, Neil Patel wrote: > Hi Moises, > > I recently added a second PRI line to my A102 and followed your > instructions below (modified /etc/wanpipe/smg_pri.conf; openzap.conf as > shown below) and can successfully dial out with g1, but not the recently > added second line set on g2. Below is the error we get: > > freeswitch at otalo> originate openzap/smg_prid/a/@g2 &echo > 2010-06-21 19:15:59.591930 [WARNING] ozmod_sangoma_boost.c:344 TX EVENT: > CALL_START:(80) [w1g1] CSid=3 Seq=4 Cn=[N/A] Cd=[9586550654] Ci=[N/A] > Rdnis=[] > 2010-06-21 19:15:59.624992 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT > (N): CALL_START_ACK:(81) [w2g1] Rc=0 CSid=3 Seq=6 > 2010-06-21 19:15:59.625987 [NOTICE] switch_channel.c:675 New Channel > OpenZAP/1:31/@g2 [19ae26fb-f9e4-480a-9cc7-11fa91b8bb1c] > 2010-06-21 19:15:59.724996 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT > (N): CALL_STOPPED:(85) [w2g1] Rc=3 CSid=3 Seq=7 > 2010-06-21 19:15:59.724996 [NOTICE] mod_openzap.c:1935 Hangup > OpenZAP/1:31/@g2 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] > > -ERR NO_ROUTE_DESTINATION > > Seems like a config issue, but not sure what I'm missing. Any ideas? > > Thanks in advance, > Neil > > On Sun, May 16, 2010 at 10:50 AM, Moises Silva wrote: > >> Hi Again Neil, >> >> I just noticed your dial string is incorrect. The correct syntax is: >> >> OpenZAP///[number] >> >> The span and chan code are mandatory. The number is optional ( FXS >> channels do not require a number, they just ring the FXO device connected to >> them). >> >> The span is either a number ( span id, the id is a number assigned in the >> order in which the span is defined in openzap.conf ) or a span name also as >> specified in the [span wanpipe ] line in openzap.conf >> >> The chan code is either a number ( for spans that support individual >> channel selection, boost is NOT one of them ), or a channel hunting mode, >> there is currently 2 modes, "a" is top down and "A" is bottom up. >> >> So, this is a valid string for you case: >> >> OpenZAP/smg_prid/a/ >> >> In the specific case of boost in socket mode ( openzap only supports >> socket mode ) the number may contain @gX where X is a group ( for hunting as >> configured in /etc/wanpipe/smg_pri.conf). Boost signaling are a special case >> because the hunting for channels is not done within FreeSWITCH but in >> sangoma_prid binary ( in the new OpenZAP version called FreeTDM this has >> changed depending on configuration). >> >> Bottom line, this should work: >> >> OpenZAP/smg_prid/a/1234 at g1 >> >> If you have g1 configured in /etc/wanpipe/smg_pri.conf >> >> Moises Silva >> Senior Software Engineer >> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R >> 9T3 Canada >> t. 1 905 474 1990 x 128 | e. moy at sangoma.com >> >> >> On Sat, May 15, 2010 at 6:26 PM, Neil Patel wrote: >> >>> Span was originally in the boost section when I got this error, so I >>> thought I'd try it in analog and both. None work. >>> >>> -Neil >>> >>> On Sat, May 15, 2010 at 3:18 PM, Moises Silva wrote: >>> >>>> Why do you have 2 spans in openzap.conf.xml with the same name, in both >>>> the boost and analog sections? >>>> Moises Silva >>>> Senior Software Engineer >>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R >>>> 9T3 Canada >>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com >>>> >>>> >>>> On Sat, May 15, 2010 at 3:52 PM, Neil Patel wrote: >>>> >>>>> Hi All, >>>>> >>>>> I am trying to dial out over my PRI, and am getting this error: >>>>> >>>>> 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED! >>>>> 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels >>>>> available >>>>> 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot >>>>> create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION] >>>>> >>>>> >>>>> This is my openzap.conf: >>>>> >>>>> [span wanpipe smg_prid] >>>>> name => smg_prid >>>>> trunk_type =>e1 >>>>> b-channel => 1:1-15 >>>>> b-channel => 1:17-31 >>>>> trunk_type =>e1 >>>>> b-channel => 2:1-15 >>>>> b-channel => 2:17-31 >>>>> >>>>> >>>>> This is my openzap.conf.xml: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> And here is the lua code I'm using to dial out: >>>>> >>>>> sessiondata = "OpenZAP/smg_prid/" >>>>> new_session = freeswitch.Session(sessiondata) >>>>> >>>>> >>>>> What am I missing here? >>>>> Thanks, >>>>> Neil >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.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 >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/c2e5d74b/attachment-0001.html From tgraziano at myitdepartment.net Mon Jun 21 05:56:12 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Mon, 21 Jun 2010 08:56:12 -0400 Subject: [Freeswitch-users] Understanding mod_shout "licensing" Message-ID: Since mod_shout encodes/decodes and audio file in the MP3 format, can someone explain to me how this complies with licensing since MP3 encode/decode software is not "free". Is the patent holder giving a free license to use the technology since it is a open source package? Is another technology being used that does not require licensing? Thanks, Tony From philipp.schrader at gmail.com Mon Jun 21 10:45:17 2010 From: philipp.schrader at gmail.com (Philipp Schrader) Date: Mon, 21 Jun 2010 13:45:17 -0400 Subject: [Freeswitch-users] Can't get freeswitch to send BYE packet on hangup Message-ID: Dear All, Loving freeswitch so far. There's one thing I need help with though. I can't seem to get freeswitch to send BYE packets to my originating call when the hangup happens. At first I tried this in Lua, but the symptoms are still the same with a barebone dialplan: I also turned siptrace on with the commands: sofia profile internal siptrace on sofia profile external siptrace on All I can get though is a SIP/2.0 200 OK at the beginning of the call, but no BYE at the end: Does anyone have an idea as to what I am missing? 2010-06-21 13:10:20.600232 [INFO] switch_cpp.cpp:1142 username: XXXX, password: XXXXXXXX 2010-06-21 13:10:20.603790 [NOTICE] switch_channel.c:669 New Channel sofia/internal/XXXX at aaaaa [6a2c61cc-4f44-4a33-b6fc- af9728c2f1d2] 2010-06-21 13:10:20.607511 [INFO] mod_dialplan_xml.c:418 Processing XXXX->103 in context default 2010-06-21 13:10:20.618949 [NOTICE] mod_dptools.c:719 Channel [sofia/internal/XXXX at aaaaa] has been answered send 1228 bytes to udp/[192.168.202.67]:5060 at 17:10:20.619691: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.202.67:5060 ;rport=5060;branch=z9hG4bKPjf2d4bf79-38fb-45fe-82bf-13b070750f90 From: sip:XXXX at aaaaa;tag=64797d89-3747-4240-abe0-28cd75241aff To: ;tag=8rcH55QDQpmDD Call-ID: 14b9b208-6690-495e-a403-ae7da6ddd07a CSeq: 24285 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 1800;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 251 Remote-Party-ID: "103" >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1277110358 1277110359 IN IP4 192.168.202.31 s=FreeSWITCH c=IN IP4 192.168.202.31 t=0 0 m=audio 29862 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ recv 386 bytes from udp/[192.168.202.67]:5060 at 17:10:20.628813: ------------------------------------------------------------------------ ACK sip:103 at 192.168.202.31:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.202.67:5060 ;rport;branch=z9hG4bKPj65f1608c-62e2-4ca8-973b-8d8daf8a9a6b Max-Forwards: 70 From: sip:XXXX at aaaaa;tag=64797d89-3747-4240-abe0-28cd75241aff To: sip:103 at aaaaa;tag=8rcH55QDQpmDD Call-ID: 14b9b208-6690-495e-a403-ae7da6ddd07a CSeq: 24285 ACK Content-Length: 0 ------------------------------------------------------------------------ 2010-06-21 13:10:25.618976 [NOTICE] mod_dptools.c:705 Hangup sofia/internal/XXXX at aaaaa [CS_EXECUTE] [NORMAL_CLEARING] 2010-06-21 13:10:25.629655 [NOTICE] switch_core_session.c:1182 Session 2 (sofia/internal/XXXX at aaaaa) Ended 2010-06-21 13:10:25.629655 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/XXXX at aaaaa [CS_DESTROY] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/3933a966/attachment.html From brian at freeswitch.org Mon Jun 21 12:25:21 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Jun 2010 14:25:21 -0500 Subject: [Freeswitch-users] Can't get freeswitch to send BYE packet on hangup In-Reply-To: References: Message-ID: type "sofia loglevel all 9" Then try again I suspect its sending it but getting an error. /b On Jun 21, 2010, at 12:45 PM, Philipp Schrader wrote: > Dear All, > > Loving freeswitch so far. > There's one thing I need help with though. From mike at jerris.com Mon Jun 21 12:39:34 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 21 Jun 2010 15:39:34 -0400 Subject: [Freeswitch-users] Understanding mod_shout "licensing" In-Reply-To: References: Message-ID: The best site I have seen in regards to mp3 licensing requirements is: http://mp3licensing.com They seem to indicate that commercial use requires a license: http://mp3licensing.com/help/index.html#4 http://mp3licensing.com/help/index.html#5 I am not a lawyer, I have no knowledge if this site represents all possible patent claims related to mp3. As always, I would advise people using technologies they suspect are patent protected to seek legal advice to understand their own obligations based on their use and legal jurisdiction. Mike On Jun 21, 2010, at 8:56 AM, Tony Graziano wrote: > Since mod_shout encodes/decodes and audio file in the MP3 format, can > someone explain to me how this complies with licensing since MP3 > encode/decode software is not "free". > > Is the patent holder giving a free license to use the technology since > it is a open source package? Is another technology being used that > does not require licensing? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/77b0367d/attachment.html From philipp.schrader at gmail.com Mon Jun 21 12:54:41 2010 From: philipp.schrader at gmail.com (Philipp Schrader) Date: Mon, 21 Jun 2010 15:54:41 -0400 Subject: [Freeswitch-users] Can't get freeswitch to send BYE packet on hangup In-Reply-To: References: Message-ID: Hi Brian, Thanks for your reply, looks like I do get an error. I'm reading it as some DNS issue, but I don't know why it's trying to reach a "_sip._udp." subdomain. Note: 192.168.202.94 is my local dns server incoming_reclaim_all((nil), (nil), 0x4023ee40) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 55 ms 2010-06-21 15:44:32.489712 [NOTICE] mod_dptools.c:705 Hangup sofia/internal/XXXX at aaaaa.AAAAA [CS_EXECUTE] [NORMAL_CLEARING] nua: nua_bye: entering nua(0x8ae9c0): sent signal r_bye 2010-06-21 15:44:32.504861 [NOTICE] switch_core_session.c:1182 Session 1 (sofia/internal/XXXX at aaaaa.AAAAA) Ended 2010-06-21 15:44:32.504861 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/XXXX at aaaaa.AAAAA [CS_DESTROY] nua(0x8ae9c0): recv signal r_bye nua: nua_stack_set_params: entering soa_set_params(static::0x8ab4f0, ...) called soa_terminate(static::0x8ab4f0) called soa_init_offer_answer(static::0x8ab4f0) called nta: selecting scheme sip sres_cache_get(0x701180, NAPTR, "aaaaa.AAAAA.") called nta: for "aaaaa.AAAAA" query "aaaaa.AAAAA" NAPTR sres_query(0x6f32d0, 0x7f25980050d0, NAPTR, "aaaaa.AAAAA") called sres_send_dns_query(0x6f32d0, 0x7f25980057b0) called sres_sofia_update(0x71ecd0, 50, -1) sres_send_dns_query(0x6f32d0, 0x7f25980057b0) id=17457 NAPTR aaaaa.AAAAA (to [192.168.202.94]:53) sres_resolver_receive(0x6f32d0, 50) called AUTHORITY RR received AAAAA. SOA IN 7247 rdlen=48 sres_resolver_receive(0x6f32d0, 0x7f25980057b0) id=17457 (from [192.168.202.94]:53) sres(q=0x7f25980057b0): reporting error NAME_ERR for NAPTR aaaaa.AAAAA sres_cache_get(0x701180, SRV, "_sip._udp.aaaaa.AAAAA.") called nta: for "aaaaa.AAAAA" query "_sip._udp.aaaaa.AAAAA" SRV sres_query(0x6f32d0, 0x7f25980050d0, SRV, "_sip._udp.aaaaa.AAAAA") called sres_send_dns_query(0x6f32d0, 0x7f2598005960) called sres_send_dns_query(0x6f32d0, 0x7f2598005960) id=17458 SRV _sip._udp.aaaaa.AAAAA (to [192.168.202.94]:53) sres_resolver_receive(0x6f32d0, 50) called AUTHORITY RR received AAAAA. SOA IN 7247 rdlen=48 sres_resolver_receive(0x6f32d0, 0x7f2598005960) id=17458 (from [192.168.202.94]:53) sres(q=0x7f2598005960): reporting error NAME_ERR for SRV _sip._udp.aaaaa.AAAAA sres_cache_get(0x701180, A, "aaaaa.AAAAA.") called nta: for "aaaaa.AAAAA" query "aaaaa.AAAAA" A sres_query(0x6f32d0, 0x7f25980050d0, A, "aaaaa.AAAAA") called sres_send_dns_query(0x6f32d0, 0x7f2598005b20) called sres_send_dns_query(0x6f32d0, 0x7f2598005b20) id=17459 A aaaaa.AAAAA (to [192.168.202.94]:53) sres_resolver_receive(0x6f32d0, 50) called AUTHORITY RR received AAAAA. SOA IN 7247 rdlen=48 sres_resolver_receive(0x6f32d0, 0x7f2598005b20) id=17459 (from [192.168.202.94]:53) sres(q=0x7f2598005b20): reporting error NAME_ERR for A aaaaa.AAAAA nua(0x8ae9c0): event r_bye 503 DNS Error nua(0x8ae9c0): call state changed: terminating -> terminated nua(0x8ae9c0): event i_state 503 to BYE nua(0x8ae9c0): event i_terminated 503 to BYE nua(0x8ae9c0): removing session usage soa_destroy(static::0x8ab4f0) called nta_leg_destroy(0x7c2850) nua: terminated session 0x8ae9c0 nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x8ae9c0): sent signal r_destroy nua(0x8ae9c0): recv signal r_destroy nta_leg_destroy((nil)) nta: timer I fired, terminate 200 response incoming_reclaim_all((nil), (nil), 0x4023ee40) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer set next to 4989 ms nta: timer K fired, terminate BYE (132456400) outgoing_reclaim_all((nil), (nil), 0x4023ee30) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set On Mon, Jun 21, 2010 at 15:25, Brian West wrote: > type "sofia loglevel all 9" > > Then try again I suspect its sending it but getting an error. > > /b > > On Jun 21, 2010, at 12:45 PM, Philipp Schrader wrote: > > > Dear All, > > > > Loving freeswitch so far. > > There's one thing I need help with though. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/7b7529d5/attachment-0001.html From brian at freeswitch.org Mon Jun 21 13:04:36 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Jun 2010 15:04:36 -0500 Subject: [Freeswitch-users] Can't get freeswitch to send BYE packet on hangup In-Reply-To: References: Message-ID: Because its trying to resolve the SRV record for the packet to send the bye and its failing. So the SRV fails as does the A record so the DNS failure results in the bye not being sent. /b On Jun 21, 2010, at 2:54 PM, Philipp Schrader wrote: > Hi Brian, > > Thanks for your reply, looks like I do get an error. > I'm reading it as some DNS issue, but I don't know why it's trying to reach a "_sip._udp." subdomain. > Note: 192.168.202.94 is my local dns server From sos at sokhapkin.dyndns.org Mon Jun 21 13:05:25 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 21 Jun 2010 16:05:25 -0400 Subject: [Freeswitch-users] Can't get freeswitch to send BYE packet on hangup In-Reply-To: References: Message-ID: <201006211605.25149.sos@sokhapkin.dyndns.org> The question is not related to FS. Learn about DNS SRV records, "_sip._udp." is not a subdomain, it's a SRV record. On Monday 21 June 2010, Philipp Schrader wrote: > Hi Brian, > > Thanks for your reply, looks like I do get an error. > I'm reading it as some DNS issue, but I don't know why it's trying to reach > a "_sip._udp." subdomain. > Note: 192.168.202.94 is my local dns server > > incoming_reclaim_all((nil), (nil), 0x4023ee40) > nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free > nta: timer set next to 55 ms > 2010-06-21 15:44:32.489712 [NOTICE] mod_dptools.c:705 Hangup > sofia/internal/XXXX at aaaaa.AAAAA [CS_EXECUTE] [NORMAL_CLEARING] > nua: nua_bye: entering > nua(0x8ae9c0): sent signal r_bye > 2010-06-21 15:44:32.504861 [NOTICE] switch_core_session.c:1182 Session 1 > (sofia/internal/XXXX at aaaaa.AAAAA) Ended > 2010-06-21 15:44:32.504861 [NOTICE] switch_core_session.c:1184 Close > Channel sofia/internal/XXXX at aaaaa.AAAAA [CS_DESTROY] > nua(0x8ae9c0): recv signal r_bye > nua: nua_stack_set_params: entering > soa_set_params(static::0x8ab4f0, ...) called > soa_terminate(static::0x8ab4f0) called > soa_init_offer_answer(static::0x8ab4f0) called > nta: selecting scheme sip > sres_cache_get(0x701180, NAPTR, "aaaaa.AAAAA.") called > nta: for "aaaaa.AAAAA" query "aaaaa.AAAAA" NAPTR > sres_query(0x6f32d0, 0x7f25980050d0, NAPTR, "aaaaa.AAAAA") called > sres_send_dns_query(0x6f32d0, 0x7f25980057b0) called > sres_sofia_update(0x71ecd0, 50, -1) > sres_send_dns_query(0x6f32d0, 0x7f25980057b0) id=17457 NAPTR aaaaa.AAAAA > (to [192.168.202.94]:53) > sres_resolver_receive(0x6f32d0, 50) called > AUTHORITY RR received AAAAA. SOA IN 7247 rdlen=48 > sres_resolver_receive(0x6f32d0, 0x7f25980057b0) id=17457 (from > [192.168.202.94]:53) > sres(q=0x7f25980057b0): reporting error NAME_ERR for NAPTR aaaaa.AAAAA > sres_cache_get(0x701180, SRV, "_sip._udp.aaaaa.AAAAA.") called > nta: for "aaaaa.AAAAA" query "_sip._udp.aaaaa.AAAAA" SRV > sres_query(0x6f32d0, 0x7f25980050d0, SRV, "_sip._udp.aaaaa.AAAAA") called > sres_send_dns_query(0x6f32d0, 0x7f2598005960) called > sres_send_dns_query(0x6f32d0, 0x7f2598005960) id=17458 SRV > _sip._udp.aaaaa.AAAAA (to [192.168.202.94]:53) > sres_resolver_receive(0x6f32d0, 50) called > AUTHORITY RR received AAAAA. SOA IN 7247 rdlen=48 > sres_resolver_receive(0x6f32d0, 0x7f2598005960) id=17458 (from > [192.168.202.94]:53) > sres(q=0x7f2598005960): reporting error NAME_ERR for SRV > _sip._udp.aaaaa.AAAAA > sres_cache_get(0x701180, A, "aaaaa.AAAAA.") called > nta: for "aaaaa.AAAAA" query "aaaaa.AAAAA" A > sres_query(0x6f32d0, 0x7f25980050d0, A, "aaaaa.AAAAA") called > sres_send_dns_query(0x6f32d0, 0x7f2598005b20) called > sres_send_dns_query(0x6f32d0, 0x7f2598005b20) id=17459 A aaaaa.AAAAA (to > [192.168.202.94]:53) > sres_resolver_receive(0x6f32d0, 50) called > AUTHORITY RR received AAAAA. SOA IN 7247 rdlen=48 > sres_resolver_receive(0x6f32d0, 0x7f2598005b20) id=17459 (from > [192.168.202.94]:53) > sres(q=0x7f2598005b20): reporting error NAME_ERR for A aaaaa.AAAAA > nua(0x8ae9c0): event r_bye 503 DNS Error > nua(0x8ae9c0): call state changed: terminating -> terminated > nua(0x8ae9c0): event i_state 503 to BYE > nua(0x8ae9c0): event i_terminated 503 to BYE > nua(0x8ae9c0): removing session usage > soa_destroy(static::0x8ab4f0) called > nta_leg_destroy(0x7c2850) > nua: terminated session 0x8ae9c0 > nua: nua_application_event: entering > nua: nua_handle_magic: entering > nua: nua_application_event: entering > nua: nua_handle_magic: entering > nua: nua_handle_bind: entering > nua: nua_application_event: entering > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nua(0x8ae9c0): sent signal r_destroy > nua(0x8ae9c0): recv signal r_destroy > nta_leg_destroy((nil)) > nta: timer I fired, terminate 200 response > incoming_reclaim_all((nil), (nil), 0x4023ee40) > nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free > nta: timer set next to 4989 ms > nta: timer K fired, terminate BYE (132456400) > outgoing_reclaim_all((nil), (nil), 0x4023ee30) > nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free > nta: timer not set > > On Mon, Jun 21, 2010 at 15:25, Brian West wrote: > > type "sofia loglevel all 9" > > > > Then try again I suspect its sending it but getting an error. > > > > /b > > > > On Jun 21, 2010, at 12:45 PM, Philipp Schrader wrote: > > > Dear All, > > > > > > Loving freeswitch so far. > > > There's one thing I need help with though. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 infos at madovsky.org Mon Jun 21 13:33:22 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 21 Jun 2010 16:33:22 -0400 Subject: [Freeswitch-users] voicemail wav filename Message-ID: I'd like to change the name of the voicemail attached. but can't find the right conf file.. anyone ? Thanks F ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 21, 2010 12:47 PM Subject: voicemail wav filename Hi, where can I customize the wav filename sent from voicemail ? Thanks F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100621/50736bf3/attachment.html From brian at freeswitch.org Mon Jun 21 13:41:59 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Jun 2010 15:41:59 -0500 Subject: [Freeswitch-users] voicemail wav filename In-Reply-To: References: Message-ID: <57812EEC-E274-4675-943E-E47E9DDAC3BF@freeswitch.org> You can't. Patches welcome. /b On Jun 21, 2010, at 3:33 PM, Madovsky wrote: > I'd like to change the name of the voicemail attached. > but can't find the right conf file.. > anyone ? > > Thanks > > F From philipp.schrader at gmail.com Mon Jun 21 14:49:15 2010 From: philipp.schrader at gmail.com (Philipp Schrader) Date: Mon, 21 Jun 2010 17:49:15 -0400 Subject: [Freeswitch-users] Can't get freeswitch to send BYE packet on hangup In-Reply-To: <201006211605.25149.sos@sokhapkin.dyndns.org> References: <201006211605.25149.sos@sokhapkin.dyndns.org> Message-ID: My apologies; you are correct. I was getting this issue in a virtual machine. On the live server the problem went away; the DNS server was able to properly resolve the record for freeswitch. Thank you very much for your help! On Mon, Jun 21, 2010 at 16:05, Sergey Okhapkin wrote: > The question is not related to FS. Learn about DNS SRV records, > "_sip._udp." > is not a subdomain, it's a SRV record. > > On Monday 21 June 2010, Philipp Schrader wrote: > > Hi Brian, > > > > Thanks for your reply, looks like I do get an error. > > I'm reading it as some DNS issue, but I don't know why it's trying to > reach > > a "_sip._udp." subdomain. > > Note: 192.168.202.94 is my local dns server > > > > incoming_reclaim_all((nil), (nil), 0x4023ee40) > > nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free > > nta: timer set next to 55 ms > > 2010-06-21 15:44:32.489712 [NOTICE] mod_dptools.c:705 Hangup > > sofia/internal/XXXX at aaaaa.AAAAA [CS_EXECUTE] [NORMAL_CLEARING] > > nua: nua_bye: entering > > nua(0x8ae9c0): sent signal r_bye > > 2010-06-21 15:44:32.504861 [NOTICE] switch_core_session.c:1182 Session 1 > > (sofia/internal/XXXX at aaaaa.AAAAA) Ended > > 2010-06-21 15:44:32.504861 [NOTICE] switch_core_session.c:1184 Close > > Channel sofia/internal/XXXX at aaaaa.AAAAA [CS_DESTROY] > > nua(0x8ae9c0): recv signal r_bye > > nua: nua_stack_set_params: entering > > soa_set_params(static::0x8ab4f0, ...) called > > soa_terminate(static::0x8ab4f0) called > > soa_init_offer_answer(static::0x8ab4f0) called > > nta: selecting scheme sip > > sres_cache_get(0x701180, NAPTR, "aaaaa.AAAAA.") called > > nta: for "aaaaa.AAAAA" query "aaaaa.AAAAA" NAPTR > > sres_query(0x6f32d0, 0x7f25980050d0, NAPTR, "aaaaa.AAAAA") called > > sres_send_dns_query(0x6f32d0, 0x7f25980057b0) called > > sres_sofia_update(0x71ecd0, 50, -1) > > sres_send_dns_query(0x6f32d0, 0x7f25980057b0) id=17457 NAPTR aaaaa.AAAAA > > (to [192.168.202.94]:53) > > sres_resolver_receive(0x6f32d0, 50) called > > AUTHORITY RR received AAAAA. SOA IN 7247 rdlen=48 > > sres_resolver_receive(0x6f32d0, 0x7f25980057b0) id=17457 (from > > [192.168.202.94]:53) > > sres(q=0x7f25980057b0): reporting error NAME_ERR for NAPTR aaaaa.AAAAA > > sres_cache_get(0x701180, SRV, "_sip._udp.aaaaa.AAAAA.") called > > nta: for "aaaaa.AAAAA" query "_sip._udp.aaaaa.AAAAA" SRV > > sres_query(0x6f32d0, 0x7f25980050d0, SRV, "_sip._udp.aaaaa.AAAAA") called > > sres_send_dns_query(0x6f32d0, 0x7f2598005960) called > > sres_send_dns_query(0x6f32d0, 0x7f2598005960) id=17458 SRV > > _sip._udp.aaaaa.AAAAA (to [192.168.202.94]:53) > > sres_resolver_receive(0x6f32d0, 50) called > > AUTHORITY RR received AAAAA. SOA IN 7247 rdlen=48 > > sres_resolver_receive(0x6f32d0, 0x7f2598005960) id=17458 (from > > [192.168.202.94]:53) > > sres(q=0x7f2598005960): reporting error NAME_ERR for SRV > > _sip._udp.aaaaa.AAAAA > > sres_cache_get(0x701180, A, "aaaaa.AAAAA.") called > > nta: for "aaaaa.AAAAA" query "aaaaa.AAAAA" A > > sres_query(0x6f32d0, 0x7f25980050d0, A, "aaaaa.AAAAA") called > > sres_send_dns_query(0x6f32d0, 0x7f2598005b20) called > > sres_send_dns_query(0x6f32d0, 0x7f2598005b20) id=17459 A aaaaa.AAAAA (to > > [192.168.202.94]:53) > > sres_resolver_receive(0x6f32d0, 50) called > > AUTHORITY RR received AAAAA. SOA IN 7247 rdlen=48 > > sres_resolver_receive(0x6f32d0, 0x7f2598005b20) id=17459 (from > > [192.168.202.94]:53) > > sres(q=0x7f2598005b20): reporting error NAME_ERR for A aaaaa.AAAAA > > nua(0x8ae9c0): event r_bye 503 DNS Error > > nua(0x8ae9c0): call state changed: terminating -> terminated > > nua(0x8ae9c0): event i_state 503 to BYE > > nua(0x8ae9c0): event i_terminated 503 to BYE > > nua(0x8ae9c0): removing session usage > > soa_destroy(static::0x8ab4f0) called > > nta_leg_destroy(0x7c2850) > > nua: terminated session 0x8ae9c0 > > nua: nua_application_event: entering > > nua: nua_handle_magic: entering > > nua: nua_application_event: entering > > nua: nua_handle_magic: entering > > nua: nua_handle_bind: entering > > nua: nua_application_event: entering > > nua: nua_handle_magic: entering > > nua: nua_handle_destroy: entering > > nua(0x8ae9c0): sent signal r_destroy > > nua(0x8ae9c0): recv signal r_destroy > > nta_leg_destroy((nil)) > > nta: timer I fired, terminate 200 response > > incoming_reclaim_all((nil), (nil), 0x4023ee40) > > nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free > > nta: timer set next to 4989 ms > > nta: timer K fired, terminate BYE (132456400) > > outgoing_reclaim_all((nil), (nil), 0x4023ee30) > > nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free > > nta: timer not set > > > > On Mon, Jun 21, 2010 at 15:25, Brian West wrote: > > > type "sofia loglevel all 9" > > > > > > Then try again I suspect its sending it but getting an error. > > > > > > /b > > > > > > On Jun 21, 2010, at 12:45 PM, Philipp Schrader wrote: > > > > Dear All, > > > > > > > > Loving freeswitch so far. > > > > There's one thing I need help with though. > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.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/20100621/2ad7ba08/attachment-0001.html From jesse at mactechs.com Mon Jun 21 16:28:33 2010 From: jesse at mactechs.com (Jesse Peterson) Date: Mon, 21 Jun 2010 16:28:33 -0700 Subject: [Freeswitch-users] Snom & SCA weirdness In-Reply-To: References: Message-ID: <7EACD65E-3A1F-4890-A68E-D26A1061F827@mactechs.com> Anthony, Thanks for your first reply. Could you be a little more specific as to what is wrong with the Snoms? Is there an issue already open with them? I'd like to give Snom very specific details. My issue with regard to this is #2010061910000014. Thanks, - Jesse On Jun 18, 2010, at 2:13 PM, Anthony Minessale wrote: > snom and SCA is broken on all firmwares, you need to ask them to fix it. > We have reported it to them already. P.S. Please Reply-All as I subscribe to the digest and may not immediately receive replies. Thank you. From brian at freeswitch.org Mon Jun 21 16:35:12 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Jun 2010 18:35:12 -0500 Subject: [Freeswitch-users] Snom & SCA weirdness In-Reply-To: <7EACD65E-3A1F-4890-A68E-D26A1061F827@mactechs.com> References: <7EACD65E-3A1F-4890-A68E-D26A1061F827@mactechs.com> Message-ID: <3EFAFB52-A39D-4ECF-9D19-84E061FD4FD8@freeswitch.org> same thing wrong with the aastra's they are missing vital info in the call-info header to tell what appearance-index the call is associated with thru out the call setup and dialog. /b On Jun 21, 2010, at 6:28 PM, Jesse Peterson wrote: > Anthony, > > Thanks for your first reply. Could you be a little more specific as to what is wrong with the Snoms? Is there an issue already open with them? I'd like to give Snom very specific details. My issue with regard to this is #2010061910000014. > > Thanks, > - Jesse From mcampbellsmith at gmail.com Mon Jun 21 16:51:44 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 22 Jun 2010 09:51:44 +1000 Subject: [Freeswitch-users] G729 Encoder Error In-Reply-To: <8DDED232-4E7A-41D0-A8EF-ABB15C755F7D@freeswitch.org> References: <8DDED232-4E7A-41D0-A8EF-ABB15C755F7D@freeswitch.org> Message-ID: Nope. I don't use Xen. But as I wrote below, restarting FS fixed the issue. /M On Mon, Jun 21, 2010 at 11:46 PM, Brian West wrote: > Are you running under Xen? > > /b > > On Jun 20, 2010, at 6:45 AM, Mark Campbell-Smith wrote: > >> Thanks Anthony. ?I actually had followed the instructions in the txt >> file, and had unloaded mod g729. >> >> Anyway, a restart of FS solved the problem (and its up to the latest >> git checkout) >> >> Cheers > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brian at freeswitch.org Mon Jun 21 17:00:25 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Jun 2010 19:00:25 -0500 Subject: [Freeswitch-users] G729 Encoder Error In-Reply-To: References: <8DDED232-4E7A-41D0-A8EF-ABB15C755F7D@freeswitch.org> Message-ID: <17F5C47F-7DA6-431E-9C03-97651A718E82@freeswitch.org> Bet the license server wasn't running usually restarting it and reload mod_com_g729 usually fixes that if the server happens to just not be there. Also reload mod_com_g729 will also restart the server. /b On Jun 21, 2010, at 6:51 PM, Mark Campbell-Smith wrote: > Nope. I don't use Xen. But as I wrote below, restarting FS fixed the issue. > > /M > > On Mon, Jun 21, 2010 at 11:46 PM, Brian West wrote: >> Are you running under Xen? >> >> /b > From mranga at gmail.com Mon Jun 21 19:02:33 2010 From: mranga at gmail.com (M. Ranganathan) Date: Mon, 21 Jun 2010 22:02:33 -0400 Subject: [Freeswitch-users] Knowing when a prompt is done playing. Message-ID: Hello, I have an application that plays a prompt in a conference. How can the application be informed when the prompt has completed playing? Thank you for any pointers. Ranga -- M. Ranganathan From nagalenoj at gmail.com Mon Jun 21 21:30:38 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Tue, 22 Jun 2010 10:00:38 +0530 Subject: [Freeswitch-users] Knowing when a prompt is done playing. In-Reply-To: References: Message-ID: You'll get some event. Have you tried listening for events in your application? On Tue, Jun 22, 2010 at 7:32 AM, M. Ranganathan wrote: > Hello, > > I have an application that plays a prompt in a conference. How can the > application be informed when the prompt has completed playing? > > Thank you for any pointers. > > Ranga > > -- > M. Ranganathan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100622/5084c633/attachment.html From abu.4000 at gmail.com Mon Jun 21 22:06:59 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Tue, 22 Jun 2010 10:36:59 +0530 Subject: [Freeswitch-users] monitor the chat In-Reply-To: References: Message-ID: Dear all, Where I can do chat configuration and all ? Thanks once again ! On Mon, Jun 21, 2010 at 4:03 PM, Abubacker siddiq wrote: > Dear all , > My requirement is to monitor the chatting of all the extensions using > outbound or inbound extensions > and I need to log those informations. > I don't know how to do this using outbound , In inbound I could not get any > events when the > chatting happens. > > Please let me know how to do this. > Thanks in advance ! > > -- > BEST REGARDS > N.ABUBACKER > SOFTWARE ENGINEER > BK SYSTEMS (P) LTD > CHENNAI-23 > -- BEST REGARDS N.ABUBACKER SOFTWARE ENGINEER BK SYSTEMS (P) LTD CHENNAI-23 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100622/fe348f1a/attachment.html From neilp at cs.stanford.edu Mon Jun 21 23:15:37 2010 From: neilp at cs.stanford.edu (Neil Patel) Date: Tue, 22 Jun 2010 06:15:37 +0000 Subject: [Freeswitch-users] how do I get more out of destination_number? Message-ID: I have two E1 PRI lines connected to FS (via Sangoma A102). When I dial the numbers, FS does not pick up the full phone number in the destination_number: for one of the lines, the last 8 digits come through, and for another line (a toll free number) only the first three digits (800). How do I get the full called number? Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100622/6305a590/attachment.html From mike at jerris.com Mon Jun 21 23:28:01 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Jun 2010 02:28:01 -0400 Subject: [Freeswitch-users] Second session with an endpoint. In-Reply-To: References: Message-ID: <9A3E0B81-A4AF-4619-B7E8-D4FB2790A767@jerris.com> can you map out a bit the big picture here? you seem to be asking a lot of very specific questions with no context whatsoever as to what you are trying to accomplish, how you are controlling freeswitch, or what the desired result is and its hard to answer things out of context. Are you doing this from a lua script, javascript, event socket, in a freeswitch module, is this part of a sip transfer, is it in a conference? There are dozens of question I don't even know to ask without much more context to what exactly you are doing. Mike On Jun 21, 2010, at 10:17 AM, M. Ranganathan wrote: > Hello, > > I want to mute one RTP session with a given endpoint and establish a > second session with that endpoint when the first is muted. Will > FreeSWITCH allow me to do this? Is a Muted session still considered to > be "active". > > Thanks > > Ranga > From mike at jerris.com Mon Jun 21 23:32:24 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Jun 2010 02:32:24 -0400 Subject: [Freeswitch-users] how do I get more out of destination_number? In-Reply-To: References: Message-ID: <1A1DDD88-E9DC-4368-886A-F28663540C52@jerris.com> you should get all the numbers sent on the actual line. This may not be all the numbers you expect, but some truncated number based on settings on the carriers switch. On Jun 22, 2010, at 2:15 AM, Neil Patel wrote: > I have two E1 PRI lines connected to FS (via Sangoma A102). When I dial the numbers, FS does not pick up the full phone number in the destination_number: for one of the lines, the last 8 digits come through, and for another line (a toll free number) only the first three digits (800). How do I get the full called number? From msc at freeswitch.org Mon Jun 21 23:39:23 2010 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 21 Jun 2010 23:39:23 -0700 Subject: [Freeswitch-users] how do I get more out of destination_number? In-Reply-To: References: Message-ID: <41AC76C3-CCB2-4088-84AC-0C5BA307FD36@freeswitch.org> This sounds like a regex issue or a carrier issue. I would recommend that you route all incoming calls to the info app in public.xml and make a few test calls. Watch the console and see if any fields contain the full dialed number. In public.xml just create a new extension with a simple condition that grabs everything: field="destination_number" expression="^(.*)$" Then a simple action application="info" and that's it. Make a call and watch the console. The call will disconnect immediately unless you put something after the info app. If you are having trouble deciphering the info output then drop it in pastebin and we will take a look. -MC Sent from my iPhone On Jun 21, 2010, at 11:15 PM, Neil Patel wrote: > I have two E1 PRI lines connected to FS (via Sangoma A102). When I > dial the numbers, FS does not pick up the full phone number in the > destination_number: for one of the lines, the last 8 digits come > through, and for another line (a toll free number) only the first > three digits (800). How do I get the full called number? > > Thanks, > Neil > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 toqeer83 at gmail.com Tue Jun 22 01:55:48 2010 From: toqeer83 at gmail.com (toqeer ali) Date: Tue, 22 Jun 2010 13:55:48 +0500 Subject: [Freeswitch-users] Freeswitch and freepbxv3 In-Reply-To: References: Message-ID: Thanks On Mon, Jun 21, 2010 at 10:04 PM, Giovanni Maruzzelli wrote: > yes it is > http://www.freepbx.org/v3/roadmap > > On Mon, Jun 21, 2010 at 6:56 PM, toqeer ali wrote: > > > > Hi all, > > > > I have a confusion... > > > > People are talking about freepbxv3 and freeswitch. But i do not > understand > > what it is.Whether it is an graphical interface for freeswitch? > > > > Please explain if someone understand this, > > > > Thanks > > > > -- > > Toqeer Ali Syed > > > > Red Hat Certified Engineer > > mob: +92 321 9059916 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100622/5f1c3bb9/attachment.html From mranga at gmail.com Tue Jun 22 02:18:35 2010 From: mranga at gmail.com (M. Ranganathan) Date: Tue, 22 Jun 2010 05:18:35 -0400 Subject: [Freeswitch-users] Knowing when a prompt is done playing. In-Reply-To: References: Message-ID: Yes I did and I do see an event. Thanks for your quick answer. Ranga. On Tue, Jun 22, 2010 at 12:30 AM, Nagalenoj H. wrote: > You'll get some event. Have you tried listening for events in your > application? > > On Tue, Jun 22, 2010 at 7:32 AM, M. Ranganathan wrote: >> >> Hello, >> >> I have an application that plays a prompt in a conference. How can the >> application be informed when the prompt has completed playing? >> >> Thank you for any pointers. >> >> Ranga >> >> -- >> M. Ranganathan >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- M. Ranganathan From mranga at gmail.com Tue Jun 22 02:27:04 2010 From: mranga at gmail.com (M. Ranganathan) Date: Tue, 22 Jun 2010 05:27:04 -0400 Subject: [Freeswitch-users] Second session with an endpoint. In-Reply-To: <9A3E0B81-A4AF-4619-B7E8-D4FB2790A767@jerris.com> References: <9A3E0B81-A4AF-4619-B7E8-D4FB2790A767@jerris.com> Message-ID: Mike, Thank you for your reply. I am trying to do the following : 1. A calls B via a back to back user agent ( not a freeswitch B2BUA ). 2. After call setup, the B2BUA needs to Invite A and B to a Free SWITCH conference. 3. After some time, the B2BUA needs to transfer A to a free SWITCH attendant (running on the same free Switch instance ) to play a prompt. Specifically this happens when A puts B on hold and B needs to be transferred to a MOH source that is written as a FS application. I am finding that even though I mute the conference in step 2 above ( I tried to mute each member of the conference in turn and also tried mute and then deaf ), there continues to be some noise that is being sent to the endpoints after step 3 thus resulting in the phone getting two media streams and hence being unable to play the MOH. So the question really is, how do I get the conference to go dead silent other than entirely killing the conference. I hope that this is a clear enough explanation. Regards, Ranga On Tue, Jun 22, 2010 at 2:28 AM, Michael Jerris wrote: > can you map out a bit the big picture here? ?you seem to be asking a lot of very specific questions with no context whatsoever as to what you are trying to accomplish, how you are controlling freeswitch, or what the desired result is and its hard to answer things out of context. ?Are you doing this from a lua script, javascript, event socket, in a freeswitch module, is this part of a sip transfer, is it in a conference? ?There are dozens of question I don't even know to ask without much more context to what exactly you are doing. > > Mike > > On Jun 21, 2010, at 10:17 AM, M. Ranganathan wrote: > >> Hello, >> >> I want to mute one RTP session with a given endpoint and establish a >> second session with that endpoint when the first is muted. Will >> FreeSWITCH allow me to do this? Is a Muted session still considered to >> be "active". >> >> Thanks >> >> Ranga >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- M. Ranganathan From Prometheus001 at gmx.net Tue Jun 22 02:47:33 2010 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 22 Jun 2010 11:47:33 +0200 Subject: [Freeswitch-users] how to get rid of second P-Asserted-Identity? In-Reply-To: <4C1F8A21.7030702@gmx.net> References: <4C1F8A21.7030702@gmx.net> Message-ID: <4C2086B5.2000703@gmx.net> Hello, I finally found the "sip_cid_type=none" setting. Best regards Peter Peter P GMX schrieb: > When receiving a call from PSTN and forwarding it to another PSTN number > and setting P-Asserted-Identity header, I found that 2 > P-Asserted-Identity headers are present in the INVITE message. As the > target provider only checks the first one, the call is denied. > > The first P-Asserted-Identity header is from the the incoming call. The > second is set in our dialplan. > Is there achance to drop the first header part? > > Best regards > Peter > > Here's the dialplan: > > > data="effective_caller_id_number=02x1204xxxxx"/> > data="effective_caller_id_name=02x1204xxxxx"/> > application="export">]]> > application="export">]]> > data="sofia/external/0162xxxxxxxxxxxxxx at provider.domain"/> > > > > Here is the corresponding INVITE. > INVITE sip:0162xxxxxxxxxxxxxx at sip1.my.domain.de SIP/2.0. > Via: SIP/2.0/UDP 82.xxx.xx.1x3:5080;rport;branch=z9hG4bKjD7UvXB8XF5jD. > Max-Forwards: 29. > From: "02x139xxxxx" ;tag=U4v0ypBB1X78F. > To: . > Call-ID: 1cdc72b9-f7ea-122d-e685-001ec9b9dd2d. > CSeq: 132448209 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15434M. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Privacy: none. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 320. > X-FS-Support: update_display. > P-Asserted-Identity: "02x139xxxxx" . > (from incoing call) > P-Asserted-Identity: > . (set in dialplan) > P-Preferred-Identity: . > . > v=0. > o=FreeSWITCH 1277106526 1277106527 IN IP4 82.xxx.xx.1x3. > s=FreeSWITCH. > c=IN IP4 82.xxx.xx.1x3. > t=0 0. > m=audio 26564 RTP/AVP 8 0 98 3 101 13. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brian at freeswitch.org Tue Jun 22 05:17:48 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Jun 2010 07:17:48 -0500 Subject: [Freeswitch-users] how do I get more out of destination_number? In-Reply-To: <41AC76C3-CCB2-4088-84AC-0C5BA307FD36@freeswitch.org> References: <41AC76C3-CCB2-4088-84AC-0C5BA307FD36@freeswitch.org> Message-ID: <0F6CEA8E-1FB7-4BED-B62E-DED6BA321004@freeswitch.org> I still for the life of me don't get how you type this long emails on that iPhone... is that painful? /b On Jun 22, 2010, at 1:39 AM, Michael S Collins wrote: > This sounds like a regex issue or a carrier issue. I would recommend > that you route all incoming calls to the info app in public.xml and > make a few test calls. Watch the console and see if any fields contain > the full dialed number. > > In public.xml just create a new extension with a simple condition that > grabs everything: > > field="destination_number" expression="^(.*)$" > > Then a simple action application="info" and that's it. Make a call and > watch the console. The call will disconnect immediately unless you > put something after the info app. If you are having trouble > deciphering the info output then drop it in pastebin and we will take > a look. > > -MC > > Sent from my iPhone From brian at freeswitch.org Tue Jun 22 05:22:17 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Jun 2010 07:22:17 -0500 Subject: [Freeswitch-users] Knowing when a prompt is done playing. In-Reply-To: References: Message-ID: <44916E6C-9D1D-4D23-BDE7-024EAA8D298B@freeswitch.org> Lets start highlighting the portions you wish to reply to and stop forwarding the three plus level response. It wastes bandwidth and is harder to follow. /b On Jun 22, 2010, at 4:18 AM, M. Ranganathan wrote: > Yes I did and I do see an event. Thanks for your quick answer. > > Ranga. From mike at jerris.com Tue Jun 22 05:34:32 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Jun 2010 08:34:32 -0400 Subject: [Freeswitch-users] Second session with an endpoint. In-Reply-To: References: <9A3E0B81-A4AF-4619-B7E8-D4FB2790A767@jerris.com> Message-ID: <73134DC2-12D6-4A7C-B293-8F66B9DE5E0C@jerris.com> "how you are controlling freeswitch" "Are you doing this from a lua script, javascript, event socket, in a freeswitch module" On Jun 22, 2010, at 5:27 AM, M. Ranganathan wrote: > Mike, > > Thank you for your reply. I am trying to do the following : > > 1. A calls B via a back to back user agent ( not a freeswitch B2BUA ). > > 2. After call setup, the B2BUA needs to Invite A and B to a Free > SWITCH conference. > > 3. After some time, the B2BUA needs to transfer A to a free SWITCH > attendant (running on the same free Switch instance ) to play a > prompt. Specifically this happens when A puts B on hold and B needs to > be transferred to a MOH source that is written as a FS application. > > I am finding that even though I mute the conference in step 2 above ( > I tried to mute each member of the conference in turn and also tried > mute and then deaf ), there continues to be some noise that is being > sent to the endpoints after step 3 thus resulting in the phone getting > two media streams and hence being unable to play the MOH. > > So the question really is, how do I get the conference to go dead > silent other than entirely killing the conference. > > I hope that this is a clear enough explanation. > > > Regards, > > Ranga > > > > > > On Tue, Jun 22, 2010 at 2:28 AM, Michael Jerris wrote: >> can you map out a bit the big picture here? you seem to be asking a lot of very specific questions with no context whatsoever as to what you are trying to accomplish, how you are controlling freeswitch, or what the desired result is and its hard to answer things out of context. Are you doing this from a lua script, javascript, event socket, in a freeswitch module, is this part of a sip transfer, is it in a conference? There are dozens of question I don't even know to ask without much more context to what exactly you are doing. >> >> Mike >> >> On Jun 21, 2010, at 10:17 AM, M. Ranganathan wrote: >> >>> Hello, >>> >>> I want to mute one RTP session with a given endpoint and establish a >>> second session with that endpoint when the first is muted. Will >>> FreeSWITCH allow me to do this? Is a Muted session still considered to >>> be "active". >>> >>> Thanks >>> >>> Ranga >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > M. Ranganathan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 mranga at gmail.com Tue Jun 22 05:42:06 2010 From: mranga at gmail.com (M. Ranganathan) Date: Tue, 22 Jun 2010 08:42:06 -0400 Subject: [Freeswitch-users] Second session with an endpoint. In-Reply-To: <73134DC2-12D6-4A7C-B293-8F66B9DE5E0C@jerris.com> References: <9A3E0B81-A4AF-4619-B7E8-D4FB2790A767@jerris.com> <73134DC2-12D6-4A7C-B293-8F66B9DE5E0C@jerris.com> Message-ID: On Tue, Jun 22, 2010 at 8:34 AM, Michael Jerris wrote: > "how you are controlling freeswitch" > "Are you doing this from a lua script, javascript, event socket, in a freeswitch module" > > Sorry I missed that last time. I am controlling it using an event socket. >From the event socket I issue commands to a conference to control it. Thanks Ranga -- M. Ranganathan From odermann at googlemail.com Tue Jun 22 06:48:49 2010 From: odermann at googlemail.com (Dennis) Date: Tue, 22 Jun 2010 15:48:49 +0200 Subject: [Freeswitch-users] Problems with "park_after_bridge" Message-ID: hi, we have noticed a problem (bug?) with the newer versions of fs. park_after_bridge is not always working as expected and as it did before (fs version before beginning of this year). we tested it with older fs versions and the fs version svn-17782 from 22.06.2010 and over the fs cli. we do the following: LEGA bgapi originate {origination_caller_id_number=1234 ,ignore_early_media=true,park_after_bridge=true,originate_timeout=0',inbound_uuid=,playback_terminators=none}sofia/gateway/test/01234567 &park() LEGB: bgapi originate {origination_caller_id_number=1234 ,ignore_early_media=true,park_after_bridge=true,originate_timeout=0',inbound_uuid=,playback_terminators=none}sofia/gateway/test/0789456123 &park() Bridge A and B uuid_bridge LEGA LEGB now, if LEGA hangs up, LEGB is not sent into park. instead LEGB is also hung up. this happens, although park_after_bridge is true. could someone help us with this, please? thanks & regards dennis From msc at freeswitch.org Tue Jun 22 08:15:20 2010 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 22 Jun 2010 08:15:20 -0700 Subject: [Freeswitch-users] how do I get more out of destination_number? In-Reply-To: <0F6CEA8E-1FB7-4BED-B62E-DED6BA321004@freeswitch.org> References: <41AC76C3-CCB2-4088-84AC-0C5BA307FD36@freeswitch.org> <0F6CEA8E-1FB7-4BED-B62E-DED6BA321004@freeswitch.org> Message-ID: I was laying in bed with my kid waiting for her to fall asleep. Yes it was painful but less so than laying there bored out of my mind. :) -MC (am now laying in my bed trying to get motivated to get up...) Sent from my iPhone On Jun 22, 2010, at 5:17 AM, Brian West wrote: > I still for the life of me don't get how you type this long emails > on that iPhone... is that painful? > > /b > > On Jun 22, 2010, at 1:39 AM, Michael S Collins wrote: > >> This sounds like a regex issue or a carrier issue. I would recommend >> that you route all incoming calls to the info app in public.xml and >> make a few test calls. Watch the console and see if any fields >> contain >> the full dialed number. >> >> In public.xml just create a new extension with a simple condition >> that >> grabs everything: >> >> field="destination_number" expression="^(.*)$" >> >> Then a simple action application="info" and that's it. Make a call >> and >> watch the console. The call will disconnect immediately unless you >> put something after the info app. If you are having trouble >> deciphering the info output then drop it in pastebin and we will take >> a look. >> >> -MC >> >> Sent from my iPhone > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 mitch.capper at gmail.com Tue Jun 22 08:35:05 2010 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 22 Jun 2010 11:35:05 -0400 Subject: [Freeswitch-users] New Skype SDK SkypeKit Message-ID: http://blogs.skype.com/devzone/2010/06/skypekit_beta.html Finally may be able to ditch the client, linux only at launch I believe. ~Mitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100622/5379878b/attachment.html From helmut.kuper at ewetel.de Tue Jun 22 08:54:12 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 22 Jun 2010 17:54:12 +0200 Subject: [Freeswitch-users] ESL: phpmod Message-ID: <4C20DCA4.7090402@ewetel.de> Hello, I have a php esl outbound deamon which is contactet by socket application when a caller hit the dialplan. The php deamon originates a call and waits after that for events. During ringing I cancel the callers call via caller's phone. Now the odd thing: Sometimes I got a HANGUP event so i can react on it, and sometimes not. In each case I can't write any command into the esl connection because it blocks endlessly. So I guess that FS has already torn down the socket in both cases. I wonder why I got no event from esl when this happens. Checking the esl connection with "connected()" returns always true resp. '1' in both cases. When I don't start an originate bgapi command, everything works and esl->connected() returns 0 resp false. Any ideas where I'm wrong? regards Helmut From gmaruzz at celliax.org Tue Jun 22 08:54:56 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 22 Jun 2010 17:54:56 +0200 Subject: [Freeswitch-users] New Skype SDK SkypeKit In-Reply-To: References: Message-ID: Very interesting, that would be a very nice basis for the next mod_skypopen... Only problem, seems it will be a "for-pay" sdk, and the deployment maybe restricted... We'll see more as things developes... -giovanni On Tue, Jun 22, 2010 at 5:35 PM, Mitch Capper wrote: > http://blogs.skype.com/devzone/2010/06/skypekit_beta.html > > Finally may be able to ditch the client, linux only at launch I believe. > > ~Mitch > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From helmut.kuper at ewetel.de Tue Jun 22 09:02:33 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 22 Jun 2010 18:02:33 +0200 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: <4C20DCA4.7090402@ewetel.de> References: <4C20DCA4.7090402@ewetel.de> Message-ID: <4C20DE99.9070005@ewetel.de> Hi, here the loop code which is not working. To get it working comment the bgapi command. [...] $agent_chan_vars="origination_caller_id_name='Queue 1',origination_caller_id_number=$caller_id_number"; $agent="2850"; $con->bgapi("originate", "[origination_uuid=$a_uuid,$agent_chan_vars]sofia/internal/$agent@$sipdomain &playback(kuper/celt.mp3)"); do { acd_log(LOG_DEBUG, "Loop $l\n"); $e = $con->recvEventTimed(1000); echo $con->connected()."\n"; } while ($con->connected()); [...] On 22.06.2010 17:54, Helmut Kuper wrote: > Hello, > > I have a php esl outbound deamon which is contactet by socket > application when a caller hit the dialplan. The php deamon originates a > call and waits after that for events. During ringing I cancel the > callers call via caller's phone. > > Now the odd thing: Sometimes I got a HANGUP event so i can react on it, > and sometimes not. In each case I can't write any command into the esl > connection because it blocks endlessly. So I guess that FS has already > torn down the socket in both cases. I wonder why I got no event from esl > when this happens. Checking the esl connection with "connected()" > returns always true resp. '1' in both cases. > > When I don't start an originate bgapi command, everything works and > esl->connected() returns 0 resp false. > > > Any ideas where I'm wrong? > > > regards > Helmut > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From sos at sokhapkin.dyndns.org Tue Jun 22 09:14:08 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 22 Jun 2010 12:14:08 -0400 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: <4C20DE99.9070005@ewetel.de> References: <4C20DCA4.7090402@ewetel.de> <4C20DE99.9070005@ewetel.de> Message-ID: <201006221214.08312.sos@sokhapkin.dyndns.org> What is the reason to use "bgapi" instead of "api"? Your code just waits for the connection completion. Also you code doesn't take into account that the connection could fail for some reason. On Tuesday 22 June 2010, Helmut Kuper wrote: > Hi, > > here the loop code which is not working. To get it working comment the > bgapi command. > > > [...] > $agent_chan_vars="origination_caller_id_name='Queue > 1',origination_caller_id_number=$caller_id_number"; > $agent="2850"; > $con->bgapi("originate", > "[origination_uuid=$a_uuid,$agent_chan_vars]sofia/internal/$agent@$sipdomai > n &playback(kuper/celt.mp3)"); > do { > acd_log(LOG_DEBUG, "Loop $l\n"); > $e = $con->recvEventTimed(1000); > echo $con->connected()."\n"; > } while ($con->connected()); > [...] > > On 22.06.2010 17:54, Helmut Kuper wrote: > > Hello, > > > > I have a php esl outbound deamon which is contactet by socket > > application when a caller hit the dialplan. The php deamon originates a > > call and waits after that for events. During ringing I cancel the > > callers call via caller's phone. > > > > Now the odd thing: Sometimes I got a HANGUP event so i can react on it, > > and sometimes not. In each case I can't write any command into the esl > > connection because it blocks endlessly. So I guess that FS has already > > torn down the socket in both cases. I wonder why I got no event from esl > > when this happens. Checking the esl connection with "connected()" > > returns always true resp. '1' in both cases. > > > > When I don't start an originate bgapi command, everything works and > > esl->connected() returns 0 resp false. > > > > > > Any ideas where I'm wrong? > > > > > > regards > > Helmut > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > From Prometheus001 at gmx.net Tue Jun 22 10:10:37 2010 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 22 Jun 2010 19:10:37 +0200 Subject: [Freeswitch-users] xml-curl through proxy Message-ID: <4C20EE8D.2070903@gmx.net> Hello, is there a way to pass xml-curl reuqest through a http proxy? For debugging puposes I would like to reroute destinct requests to another destination. Best regards Peter From helmut.kuper at ewetel.de Tue Jun 22 10:44:02 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 22 Jun 2010 19:44:02 +0200 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: <201006221214.08312.sos@sokhapkin.dyndns.org> References: <4C20DCA4.7090402@ewetel.de> <4C20DE99.9070005@ewetel.de> <201006221214.08312.sos@sokhapkin.dyndns.org> Message-ID: <4C20F662.8020504@ewetel.de> Hi Sergey, On 22.06.2010 18:14, Sergey Okhapkin wrote: > What is the reason to use "bgapi" instead of "api"? Your code just waits for > the connection completion. Well, bgapi allows my code *not* to wait until originate connected or fails. Instead I want to get events about connect or failure... > > Also you code doesn't take into account that the connection could fail for > some reason. yes, I know, the code sippet is only for demonstrating the problem. regards helmut From sos at sokhapkin.dyndns.org Tue Jun 22 11:02:56 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 22 Jun 2010 14:02:56 -0400 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: <4C20F662.8020504@ewetel.de> References: <4C20DCA4.7090402@ewetel.de> <201006221214.08312.sos@sokhapkin.dyndns.org> <4C20F662.8020504@ewetel.de> Message-ID: <201006221402.56833.sos@sokhapkin.dyndns.org> Why not analyze the return code of "api" call? Without events loop? On Tuesday 22 June 2010, Helmut Kuper wrote: > Hi Sergey, > > On 22.06.2010 18:14, Sergey Okhapkin wrote: > > What is the reason to use "bgapi" instead of "api"? Your code just waits > > for the connection completion. > > Well, bgapi allows my code *not* to wait until originate connected or > fails. Instead I want to get events about connect or failure... > > > Also you code doesn't take into account that the connection could fail > > for some reason. > > yes, I know, the code sippet is only for demonstrating the problem. > > > regards > helmut > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 helmut.kuper at ewetel.de Tue Jun 22 11:34:04 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 22 Jun 2010 20:34:04 +0200 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: <201006221402.56833.sos@sokhapkin.dyndns.org> References: <4C20DCA4.7090402@ewetel.de> <201006221214.08312.sos@sokhapkin.dyndns.org> <4C20F662.8020504@ewetel.de> <201006221402.56833.sos@sokhapkin.dyndns.org> Message-ID: <4C21021C.5040309@ewetel.de> Hi, hm well, yes, indeed that works, too. Maybe even better. But I still have the problem of a hanging command which doesn't comes back until timeout, answer, cancel, ... on callee's side. I need to have two channels controlled. The originating caller channel and the 2nd channel originated by my deamon. I have to playback some files to caller while searching for a free callee. Once callee has answered the call the deamon bridges both channels. On 22.06.2010 20:02, Sergey Okhapkin wrote: > Why not analyze the return code of "api" call? Without events loop? > > On Tuesday 22 June 2010, Helmut Kuper wrote: >> Hi Sergey, >> >> On 22.06.2010 18:14, Sergey Okhapkin wrote: >>> What is the reason to use "bgapi" instead of "api"? Your code just waits >>> for the connection completion. >> >> Well, bgapi allows my code *not* to wait until originate connected or >> fails. Instead I want to get events about connect or failure... >> >>> Also you code doesn't take into account that the connection could fail >>> for some reason. >> >> yes, I know, the code sippet is only for demonstrating the problem. >> >> >> regards >> helmut From helmut.kuper at ewetel.de Tue Jun 22 12:01:50 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 22 Jun 2010 21:01:50 +0200 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: <4C21021C.5040309@ewetel.de> References: <4C20DCA4.7090402@ewetel.de> <201006221214.08312.sos@sokhapkin.dyndns.org> <4C20F662.8020504@ewetel.de> <201006221402.56833.sos@sokhapkin.dyndns.org> <4C21021C.5040309@ewetel.de> Message-ID: <4C21089E.4030800@ewetel.de> As a workaround I armed caller's channel with api_hangup_hook which does a uuid_kill on the b-leg. I wonder why the esl socket is blocking although I use bgapi for originate ... On 22.06.2010 20:34, Helmut Kuper wrote: > Hi, > > hm well, yes, indeed that works, too. Maybe even better. > > But I still have the problem of a hanging command which doesn't comes > back until timeout, answer, cancel, ... on callee's side. > > I need to have two channels controlled. The originating caller channel > and the 2nd channel originated by my deamon. I have to playback some > files to caller while searching for a free callee. Once callee has > answered the call the deamon bridges both channels. From sos at sokhapkin.dyndns.org Tue Jun 22 12:02:49 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 22 Jun 2010 15:02:49 -0400 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: <4C21021C.5040309@ewetel.de> References: <4C20DCA4.7090402@ewetel.de> <201006221402.56833.sos@sokhapkin.dyndns.org> <4C21021C.5040309@ewetel.de> Message-ID: <201006221502.49426.sos@sokhapkin.dyndns.org> Do not use async "bg" commands. Use their sync counterparts. Anyway you need to complete one command before issuing the next one. What you are doing now looks like event-based GUI window programming style. Switch your mind to procedural-style actions sequence programing model which is more suitable for the task you need to accomplish. On Tuesday 22 June 2010, Helmut Kuper wrote: > Hi, > > hm well, yes, indeed that works, too. Maybe even better. > > But I still have the problem of a hanging command which doesn't comes > back until timeout, answer, cancel, ... on callee's side. > > I need to have two channels controlled. The originating caller channel > and the 2nd channel originated by my deamon. I have to playback some > files to caller while searching for a free callee. Once callee has > answered the call the deamon bridges both channels. > > On 22.06.2010 20:02, Sergey Okhapkin wrote: > > Why not analyze the return code of "api" call? Without events loop? > > > > On Tuesday 22 June 2010, Helmut Kuper wrote: > >> Hi Sergey, > >> > >> On 22.06.2010 18:14, Sergey Okhapkin wrote: > >>> What is the reason to use "bgapi" instead of "api"? Your code just > >>> waits for the connection completion. > >> > >> Well, bgapi allows my code *not* to wait until originate connected or > >> fails. Instead I want to get events about connect or failure... > >> > >>> Also you code doesn't take into account that the connection could fail > >>> for some reason. > >> > >> yes, I know, the code sippet is only for demonstrating the problem. > >> > >> > >> regards > >> helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Tue Jun 22 10:58:34 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Jun 2010 10:58:34 -0700 Subject: [Freeswitch-users] Matching CHANNEL_CREATE event to previous originate command In-Reply-To: References: Message-ID: Actually, you need to subscribe to BACKGROUND_JOB events so that you can match up the uuid of the call leg with the Job-UUID. Here is a copy and paste from an actual session I just did, with what I typed in bold: <291>:*telnet localhost 8021* Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted *event plain BACKGROUND_JOB* Content-Type: command/reply Reply-Text: +OK event listener enabled plain *bgapi originate user/1001 &park()* Content-Type: command/reply Reply-Text: +OK Job-UUID: d601515b-e986-4d56-90f4-101f1a3c279f Job-UUID: d601515b-e986-4d56-90f4-101f1a3c279f Content-Length: 581 Content-Type: text/event-plain Event-Name: BACKGROUND_JOB Core-UUID: 4a408781-c3ec-4cdd-a8e0-ad13915711ba FreeSWITCH-Hostname: freeswitch1.yt FreeSWITCH-IPv4: 10.15.0.91 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2010-06-22%2010%3A51%3A50 Event-Date-GMT: Tue,%2022%20Jun%202010%2017%3A51%3A50%20GMT Event-Date-Timestamp: 1277229110556854 Event-Calling-File: mod_event_socket.c Event-Calling-Function: api_exec Event-Calling-Line-Number: 1374 Job-UUID: d601515b-e986-4d56-90f4-101f1a3c279f Job-Command: originate Job-Command-Arg: user/1001%20%26park() Content-Length: 41 +OK b5d8310a-1085-4ea5-8927-398e9eee0c39 Notice that the Job-UUID that you got in response to the bgapi command corresponds with the Job-UUID of the BACKGROUND_JOB event. The information you want, which is the result of the originate command, is found at the end of the event after "Content-Length" header. In this case the originate was successful ("+OK") and the call leg's UUID is b5d8310a... Have fun! -MC 2010/6/19 Roland H?nel > Hello everyone, > > I'm using mod_event_socket in inbound mode to 'remote control' a FreeSwitch > instance. I'm using the raw TCP interface (not any of the available language > bindings). My problem now is, if I do a command like > > bgapi originate sofia/gateway/mygw/0123456789 &park > > and try to match the channel which is created by this command to the > command itself, this seems to be very difficult. I cannot do it on timing > only, because many channels may be created in a very short time and the > order of originate-CHANNEL_CREATE will probably not be consistent. > > Is there any 'best practice' how to solve this problem? > > One solution seems to be to set {origination_uuid=x-y-z-k} in the originate > section, but this does not reliably work with failover routing (for example, > if I specify two gateways and the call gets rejected on the first and then > originated on the second, the origination_uuid will not be the same any > more). > > Greetings, > Roland > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100622/5e7d5eae/attachment.html From msc at freeswitch.org Tue Jun 22 11:41:39 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Jun 2010 11:41:39 -0700 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: <4C21021C.5040309@ewetel.de> References: <4C20DCA4.7090402@ewetel.de> <201006221214.08312.sos@sokhapkin.dyndns.org> <4C20F662.8020504@ewetel.de> <201006221402.56833.sos@sokhapkin.dyndns.org> <4C21021C.5040309@ewetel.de> Message-ID: On Tue, Jun 22, 2010 at 11:34 AM, Helmut Kuper wrote: > Hi, > > hm well, yes, indeed that works, too. Maybe even better. > > But I still have the problem of a hanging command which doesn't comes > back until timeout, answer, cancel, ... on callee's side. > Which command doesn't come back? I'm assuming you mean the "originate" itself and not the bgapi. I don't see how you can avoid a "hang" while you wait for the 2nd channel (the callee) to be contacted since you have to wait for him/her to answer. In any case, this should be quite doable either with api or with bgapi. The difference with bgapi is that you can do other stuff in your script (if you need to) while api will wait for the result of the originate before it returns. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100622/4ec1d7a6/attachment.html From djbinter at gmail.com Tue Jun 22 21:44:46 2010 From: djbinter at gmail.com (DJB International) Date: Tue, 22 Jun 2010 21:44:46 -0700 Subject: [Freeswitch-users] Dialplan failover question Message-ID: It's always confused me. I wonder whether there are any differences between these 2 dialplans. -and- What is the best way to use it? Thank you, Dorn B. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100622/b57c7262/attachment.html From helmut.kuper at ewetel.de Tue Jun 22 23:57:16 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 23 Jun 2010 08:57:16 +0200 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: References: <4C20DCA4.7090402@ewetel.de> <201006221214.08312.sos@sokhapkin.dyndns.org> <4C20F662.8020504@ewetel.de> <201006221402.56833.sos@sokhapkin.dyndns.org> <4C21021C.5040309@ewetel.de> Message-ID: <4C21B04C.3030904@ewetel.de> Hi Michael, On 22.06.2010 20:41, Michael Collins wrote: > On Tue, Jun 22, 2010 at 11:34 AM, Helmut Kuper wrote: > >> Hi, >> >> hm well, yes, indeed that works, too. Maybe even better. >> >> But I still have the problem of a hanging command which doesn't comes >> back until timeout, answer, cancel, ... on callee's side. >> > > Which command doesn't come back? I'm assuming you mean the "originate" > itself and not the bgapi. I don't see how you can avoid a "hang" while you > wait for the 2nd channel (the callee) to be contacted since you have to wait > for him/her to answer. In any case, this should be quite doable either with > api or with bgapi. The difference with bgapi is that you can do other stuff > in your script (if you need to) while api will wait for the result of the > originate before it returns. What I currently do is this: 1. caller calls in 2. FS calls my deamon via socket app (async full) 3. Deamon connects the call via ESLconnection 4. executing answer 5. executing playback 6. subscribing to all events 7. originate b-leg via bgapi 9. waiting for events (this is a loop using esl->recvEventsTimed(1000)) 9.1 replaying caller's playback 9.2 checking for caller's Hangup 9.3 checking for esl->connected=false 9.4 checking for b-leg's result (Answer or deny) 9.4.1 bridge both legs 9.4.2 release esl-socket 9.4.3 done When caller hangs up somewhere after step 7 I got sometimes a HANGUP event but more often nothing. Trying to write into the esl->socket then seems to block until originate has finished. I guess FS has torn down the socket half way or so in this state. regards Helmut From helmut.kuper at ewetel.de Wed Jun 23 01:26:49 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 23 Jun 2010 10:26:49 +0200 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: <4C21B04C.3030904@ewetel.de> References: <4C20DCA4.7090402@ewetel.de> <201006221214.08312.sos@sokhapkin.dyndns.org> <4C20F662.8020504@ewetel.de> <201006221402.56833.sos@sokhapkin.dyndns.org> <4C21021C.5040309@ewetel.de> <4C21B04C.3030904@ewetel.de> Message-ID: <4C21C549.7070700@ewetel.de> Hi again, I added now an inbound esl socket back to FS through which I execute the originate command. And this way it works. When caller hang up during originate ringing I get the HANGUP event and I can uuid_kill the originate session. :) hmmm... So bgapi originate seems so stick with the channel's outbound socket when it is called via outbound socket. regards helmut On 23.06.2010 08:57, Helmut Kuper wrote: > Hi Michael, > > On 22.06.2010 20:41, Michael Collins wrote: >> On Tue, Jun 22, 2010 at 11:34 AM, Helmut Kuper wrote: >> >>> Hi, >>> >>> hm well, yes, indeed that works, too. Maybe even better. >>> >>> But I still have the problem of a hanging command which doesn't comes >>> back until timeout, answer, cancel, ... on callee's side. >>> >> >> Which command doesn't come back? I'm assuming you mean the "originate" >> itself and not the bgapi. I don't see how you can avoid a "hang" while you >> wait for the 2nd channel (the callee) to be contacted since you have to wait >> for him/her to answer. In any case, this should be quite doable either with >> api or with bgapi. The difference with bgapi is that you can do other stuff >> in your script (if you need to) while api will wait for the result of the >> originate before it returns. > > > What I currently do is this: > > 1. caller calls in > 2. FS calls my deamon via socket app (async full) > 3. Deamon connects the call via ESLconnection > 4. executing answer > 5. executing playback > 6. subscribing to all events > 7. originate b-leg via bgapi > 9. waiting for events (this is a loop using esl->recvEventsTimed(1000)) > 9.1 replaying caller's playback > 9.2 checking for caller's Hangup > 9.3 checking for esl->connected=false > 9.4 checking for b-leg's result (Answer or deny) > 9.4.1 bridge both legs > 9.4.2 release esl-socket > 9.4.3 done > > > When caller hangs up somewhere after step 7 I got sometimes a HANGUP > event but more often nothing. Trying to write into the esl->socket then > seems to block until originate has finished. > > I guess FS has torn down the socket half way or so in this state. > > > regards > Helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 mcampbellsmith at gmail.com Wed Jun 23 05:07:08 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 23 Jun 2010 22:07:08 +1000 Subject: [Freeswitch-users] Number of codecs offerred in SDP Message-ID: Test Setup: vars.xml: The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here. FS only offers G729 and PCMU to 1020. How do I change the number of codecs that are offered to an extension? I know I can change the order in the codec_prefs, but would prefer FS to offer all three codecs to an extension. m=audio 23662 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30 Thanks From sos at sokhapkin.dyndns.org Wed Jun 23 05:17:15 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 23 Jun 2010 08:17:15 -0400 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: Message-ID: <201006230817.15506.sos@sokhapkin.dyndns.org> in SIP profile settings. On Wednesday 23 June 2010, Mark Campbell-Smith wrote: > Test Setup: > > vars.xml: > > > > The call setup is extension 1000 calls extension 1020 > 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by > FS as the A-leg codec > 2. Extension 1020 only supports GSM codec. The call fails with Not > Acceptable Here. > > FS only offers G729 and PCMU to 1020. How do I change the number of > codecs that are offered to an extension? I know I can change the > order in the codec_prefs, but would prefer FS to offer all three > codecs to an extension. > > m=audio 23662 RTP/AVP 0 18 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:30 > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sos at sokhapkin.dyndns.org Wed Jun 23 05:20:30 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 23 Jun 2010 08:20:30 -0400 Subject: [Freeswitch-users] Dialplan failover question In-Reply-To: References: Message-ID: <201006230820.30746.sos@sokhapkin.dyndns.org> I prefer the first way, it allows to set leg A channel variables to different values depending on which gateway completed the call. On Wednesday 23 June 2010, DJB International wrote: > It's always confused me. I wonder whether there are any differences > between these 2 dialplans. > > > > > > > > > > > -and- > > > > > > data="sofia/gateway/carrier1/$1|sofia/gateway/carrier2/$1"/> > > > > What is the best way to use it? > > Thank you, > Dorn B. > From david.ponzone at gmail.com Wed Jun 23 05:25:54 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Wed, 23 Jun 2010 14:25:54 +0200 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: Message-ID: <1626F217-B975-4B4E-B9EA-DB4D4FF63C2E@gmail.com> Mark, that's not normal. FS should offer PCMU/G729/GSM to 1020. Can you check your SIP Profile parameters ? Is it configured with: What FS version are you running ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/06/2010 ? 14:07, Mark Campbell-Smith a ?crit : > Test Setup: > > vars.xml: > > > > The call setup is extension 1000 calls extension 1020 > 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by > FS as the A-leg codec > 2. Extension 1020 only supports GSM codec. The call fails with Not > Acceptable Here. > > FS only offers G729 and PCMU to 1020. How do I change the number of > codecs that are offered to an extension? I know I can change the > order in the codec_prefs, but would prefer FS to offer all three > codecs to an extension. > > m=audio 23662 RTP/AVP 0 18 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:30 > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/82c54cfd/attachment.html From tgraziano at myitdepartment.net Wed Jun 23 05:28:22 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Wed, 23 Jun 2010 08:28:22 -0400 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: Message-ID: I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? Wouldn't fs only get involved when its media server is referred to? If the "other endpoint" will only accept G729, doesn't that mean you need to change that endpoint to also accept G711 or also license G729 in FS? On 6/23/10, Mark Campbell-Smith wrote: > Test Setup: > > vars.xml: > > > > The call setup is extension 1000 calls extension 1020 > 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by > FS as the A-leg codec > 2. Extension 1020 only supports GSM codec. The call fails with Not > Acceptable Here. > > FS only offers G729 and PCMU to 1020. How do I change the number of > codecs that are offered to an extension? I know I can change the > order in the codec_prefs, but would prefer FS to offer all three > codecs to an extension. > > m=audio 23662 RTP/AVP 0 18 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:30 > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. From david.ponzone at gmail.com Wed Jun 23 05:31:20 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Wed, 23 Jun 2010 14:31:20 +0200 Subject: [Freeswitch-users] Dialplan failover question In-Reply-To: <201006230820.30746.sos@sokhapkin.dyndns.org> References: <201006230820.30746.sos@sokhapkin.dyndns.org> Message-ID: Sergey, I think you can also do that with the | (pipe) syntax, using [] instead of {}. Personnaly, I would say syntax 1 is far more readable. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/06/2010 ? 14:20, Sergey Okhapkin a ?crit : > I prefer the first way, it allows to set leg A channel variables to > different > values depending on which gateway completed the call. > > On Wednesday 23 June 2010, DJB International wrote: >> It's always confused me. I wonder whether there are any differences >> between these 2 dialplans. >> >> >> >> >> >> >> >> >> >> >> -and- >> >> >> >> >> >> > data="sofia/gateway/carrier1/$1|sofia/gateway/carrier2/$1"/> >> >> >> >> What is the best way to use it? >> >> Thank you, >> Dorn B. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/29bc1360/attachment-0001.html From mcampbellsmith at gmail.com Wed Jun 23 05:43:25 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 23 Jun 2010 22:43:25 +1000 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: Message-ID: Check this good wiki page for how FS negotiates codecs (early negotiation default): http://wiki.freeswitch.org/wiki/Codec_Negotiation I have this set in my internal profile: and as stated before vars.xml: Setting late negotiation works (thanks Sergey), but reading the wiki page, I see the following sentence, which I interpret that GSM should still be sent: When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top . If B does not accept any of the codecs, the calls fails, obviously. On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano wrote: > I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? > Wouldn't fs only get involved when its media server is referred to? > > If the "other endpoint" will only accept G729, doesn't that mean you > need to change that endpoint to also accept G711 or also license G729 > in FS? > > On 6/23/10, Mark Campbell-Smith wrote: >> Test Setup: >> >> vars.xml: >> ? >> ? >> >> The call setup is extension 1000 calls extension 1020 >> 1. Extension 1000 calls with preferred codec PCMU. ?PCMU is chosen by >> FS as the A-leg codec >> 2. Extension 1020 only supports GSM codec. ?The call fails with Not >> Acceptable Here. >> >> FS only offers G729 and PCMU to 1020. ?How do I change the number of >> codecs that are offered to an extension? ?I know I can change the >> order in the codec_prefs, but would prefer FS to offer all three >> codecs to an extension. >> >> ? ?m=audio 23662 RTP/AVP 0 18 101 13 >> ? ?a=rtpmap:0 PCMU/8000 >> ? ?a=rtpmap:18 G729/8000 >> ? ?a=rtpmap:101 telephone-event/8000 >> ? ?a=fmtp:101 0-16 >> ? ?a=rtpmap:13 CN/8000 >> ? ?a=ptime:30 >> >> Thanks >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- > Sent from my mobile device > > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgraziano at voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpdesk at voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sos at sokhapkin.dyndns.org Wed Jun 23 05:49:42 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 23 Jun 2010 08:49:42 -0400 Subject: [Freeswitch-users] Dialplan failover question In-Reply-To: References: <201006230820.30746.sos@sokhapkin.dyndns.org> Message-ID: <201006230849.42240.sos@sokhapkin.dyndns.org> [] sets variables on B leg, but not on A leg. On Wednesday 23 June 2010, David Ponzone wrote: > Sergey, > > I think you can also do that with the | (pipe) syntax, using [] > instead of {}. > > Personnaly, I would say syntax 1 is far more readable. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > ? l'intention exclusive de ses destinataires. Toute utilisation ou > diffusion non autoris?e est interdite. Tout message ?lectronique est > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si > vous n'?tes pas destinataire de ce message, merci de le d?truire > imm?diatement et d'avertir l'exp?diteur. > > Le 23/06/2010 ? 14:20, Sergey Okhapkin a ?crit : > > I prefer the first way, it allows to set leg A channel variables to > > different > > values depending on which gateway completed the call. > > > > On Wednesday 23 June 2010, DJB International wrote: > >> It's always confused me. I wonder whether there are any differences > >> between these 2 dialplans. > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> -and- > >> > >> > >> > >> > >> > >> >> data="sofia/gateway/carrier1/$1|sofia/gateway/carrier2/$1"/> > >> > >> > >> > >> What is the best way to use it? > >> > >> Thank you, > >> Dorn B. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 david.ponzone at gmail.com Wed Jun 23 05:59:32 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Wed, 23 Jun 2010 14:59:32 +0200 Subject: [Freeswitch-users] Dialplan failover question In-Reply-To: <201006230849.42240.sos@sokhapkin.dyndns.org> References: <201006230820.30746.sos@sokhapkin.dyndns.org> <201006230849.42240.sos@sokhapkin.dyndns.org> Message-ID: You're absolutely right, I misread your post. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/06/2010 ? 14:49, Sergey Okhapkin a ?crit : > [] sets variables on B leg, but not on A leg. > > On Wednesday 23 June 2010, David Ponzone wrote: >> Sergey, >> >> I think you can also do that with the | (pipe) syntax, using [] >> instead of {}. >> >> Personnaly, I would say syntax 1 is far more readable. >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis >> ? l'intention exclusive de ses destinataires. Toute utilisation ou >> diffusion non autoris?e est interdite. Tout message ?lectronique >> est >> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au >> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si >> vous n'?tes pas destinataire de ce message, merci de le d?truire >> imm?diatement et d'avertir l'exp?diteur. >> >> Le 23/06/2010 ? 14:20, Sergey Okhapkin a ?crit : >>> I prefer the first way, it allows to set leg A channel variables to >>> different >>> values depending on which gateway completed the call. >>> >>> On Wednesday 23 June 2010, DJB International wrote: >>>> It's always confused me. I wonder whether there are any >>>> differences >>>> between these 2 dialplans. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> -and- >>>> >>>> >>>> >>>> >>>> >>>> >>> data="sofia/gateway/carrier1/$1|sofia/gateway/carrier2/$1"/> >>>> >>>> >>>> >>>> What is the best way to use it? >>>> >>>> Thank you, >>>> Dorn B. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/e62d5504/attachment.html From brian at freeswitch.org Wed Jun 23 06:03:59 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Jun 2010 08:03:59 -0500 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: Message-ID: Check to see if you have mod_spandsp or mod_voipcodecs loaded. /b On Jun 23, 2010, at 7:43 AM, Mark Campbell-Smith wrote: > Check this good wiki page for how FS negotiates codecs (early > negotiation default): > http://wiki.freeswitch.org/wiki/Codec_Negotiation > > I have this set in my internal profile: > > > and as stated before vars.xml: > > > > Setting late negotiation works (thanks Sergey), but reading the wiki > page, I see the following sentence, which I interpret that GSM should > still be sent: > When FS calls leg B, the list of codecs in outbound-codec-prefs for > the SIP profile is reorganized by pushing the codec negotiated above > for leg A at the top . If B does not accept any of the codecs, the > calls fails, obviously. > From david.ponzone at gmail.com Wed Jun 23 06:05:54 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Wed, 23 Jun 2010 15:05:54 +0200 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: Message-ID: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> Mark, I confirm that, as I wrote that wiki page (the early negotiation part) :) Can you really confirm your FS version ? The parameter you showed is old. codec-prefs has been replaced in SIP profiles by: David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/06/2010 ? 14:43, Mark Campbell-Smith a ?crit : > Check this good wiki page for how FS negotiates codecs (early > negotiation default): > http://wiki.freeswitch.org/wiki/Codec_Negotiation > > I have this set in my internal profile: > > > and as stated before vars.xml: > > > > Setting late negotiation works (thanks Sergey), but reading the wiki > page, I see the following sentence, which I interpret that GSM should > still be sent: > When FS calls leg B, the list of codecs in outbound-codec-prefs for > the SIP profile is reorganized by pushing the codec negotiated above > for leg A at the top . If B does not accept any of the codecs, the > calls fails, obviously. > > > > On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano > wrote: >> I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? >> Wouldn't fs only get involved when its media server is referred to? >> >> If the "other endpoint" will only accept G729, doesn't that mean you >> need to change that endpoint to also accept G711 or also license G729 >> in FS? >> >> On 6/23/10, Mark Campbell-Smith wrote: >>> Test Setup: >>> >>> vars.xml: >>> >>> >> data="outbound_codec_prefs=G729,PCMU,GSM"/> >>> >>> The call setup is extension 1000 calls extension 1020 >>> 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen >>> by >>> FS as the A-leg codec >>> 2. Extension 1020 only supports GSM codec. The call fails with Not >>> Acceptable Here. >>> >>> FS only offers G729 and PCMU to 1020. How do I change the number of >>> codecs that are offered to an extension? I know I can change the >>> order in the codec_prefs, but would prefer FS to offer all three >>> codecs to an extension. >>> >>> m=audio 23662 RTP/AVP 0 18 101 13 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:18 G729/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=rtpmap:13 CN/8000 >>> a=ptime:30 >>> >>> Thanks >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> -- >> Sent from my mobile device >> >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: tgraziano at voice.myitdepartment.net >> Fax: 434.984.8431 >> >> Email: tgraziano at myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: helpdesk at voice.myitdepartment.net >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> http://www.myitdepartment.net/gethelp/ >> >> Why do mathematicians always confuse Halloween and Christmas? >> Because 31 Oct = 25 Dec. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100623/21eb0ede/attachment.html From tgraziano at myitdepartment.net Wed Jun 23 06:19:42 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Wed, 23 Jun 2010 09:19:42 -0400 Subject: [Freeswitch-users] Understanding mod_shout "licensing" In-Reply-To: References: Message-ID: As such, is there a peer review board in FS of any sort to keep potential plugins out of the project unless they properly comply with copyright and patent laws? If not, why? The G729 plugin has a license function. Why wouldn't something like MP3? What is the benefit to FS of accepting a plugin from anyone that is operating without proper licensing from a copyright or patent holder? It seems like a slippery slope to me. I feel these are legitimate questions, and was wondering if there was a centralized license function so licenses for things like G729 and MP3 (or anything else) could be administered and offer to allow a firm to operate a legitimate server. Thanks, Tony On Mon, Jun 21, 2010 at 3:39 PM, Michael Jerris wrote: > The best site I have seen in regards to mp3 licensing requirements is: > http://mp3licensing.com > They seem to indicate that commercial use requires a license: > http://mp3licensing.com/help/index.html#4 > http://mp3licensing.com/help/index.html#5 > I am not a lawyer, I have no knowledge if this site represents all possible > patent claims related to mp3. ?As always, I would advise people using > technologies they suspect are patent protected to ?seek legal advice to > understand their own obligations based on their use and legal jurisdiction. > Mike > On Jun 21, 2010, at 8:56 AM, Tony Graziano wrote: > > Since mod_shout encodes/decodes and audio file in the MP3 format, can > someone explain to me how this complies with licensing since MP3 > encode/decode software is not "free". > > Is the patent holder giving a free license to use the technology since > it is a open source package? Is another technology being used that > does not require licensing? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. From helmut.kuper at ewetel.de Wed Jun 23 06:25:50 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 23 Jun 2010 15:25:50 +0200 Subject: [Freeswitch-users] esl phpmod background_job events not complete Message-ID: <4C220B5E.1030802@ewetel.de> Hello I found that when I sent a non existent api command via bgapi. The corresponding background_job event isn't complete: Here my command: bgapi originated ---------------------------------- Here is what I got from esl phpmod debug: [DEBUG] src/esl.c:923 esl_recv_event() RECV HEADER [Content-Length] = [528] [DEBUG] src/esl.c:923 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Name] = [BACKGROUND_JOB] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Core-UUID] = [1921f8f2-7df3-11df-90f2-7933374406ce] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [FreeSWITCH-Hostname] = [ippbx-prod] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv4] = [85.16.246.5] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [FreeSWITCH-IPv6] = [::1] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Date-Local] = [2010-06-23 15:15:36] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Date-GMT] = [Wed, 23 Jun 2010 13:15:36 GMT] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Date-Timestamp] = [1277298936734270] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Calling-File] = [mod_commands.c] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Calling-Function] = [bgapi_exec] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Event-Calling-Line-Number] = [3030] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Job-UUID] = [6977120c-7ec9-11df-a323-7933374406ce] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Job-Command] = [originated] [DEBUG] src/esl.c:1012 esl_recv_event() RECV INNER HEADER [Content-Length] = [31] [DEBUG] src/esl.c:1037 esl_recv_event() RECV EVENT Event-Name: BACKGROUND_JOB Core-UUID: 1921f8f2-7df3-11df-90f2-7933374406ce FreeSWITCH-Hostname: ippbx-prod FreeSWITCH-IPv4: 85.16.246.5 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2010-06-23 15:15:36 Event-Date-GMT: Wed, 23 Jun 2010 13:15:36 GMT Event-Date-Timestamp: 1277298936734270 Event-Calling-File: mod_commands.c Event-Calling-Function: bgapi_exec Event-Calling-Line-Number: 3030 Job-UUID: 6977120c-7ec9-11df-a323-7933374406ce Job-Command: originated Content-Length: 31 [DEBUG] src/esl.c:1045 esl_recv_event() RECV MESSAGE ---------------------------------- And here is what I got from tcpdump: Content-Length: 528 Content-Type: text/event-plain Event-Name: BACKGROUND_JOB Core-UUID: 1921f8f2-7df3-11df-90f2-7933374406ce FreeSWITCH-Hostname: ippbx-prod FreeSWITCH-IPv4: 85.16.246.5 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2010-06-23%2015%3A15%3A36 Event-Date-GMT: Wed,%2023%20Jun%202010%2013%3A15%3A36%20GMT Event-Date-Timestamp: 1277298936734270 Event-Calling-File: mod_commands.c Event-Calling-Function: bgapi_exec Event-Calling-Line-Number: 3030 Job-UUID: 6977120c-7ec9-11df-a323-7933374406ce Job-Command: originated Content-Length: 31 originated: Command not found! ---------------------------------- So, obviously the content is missing resp not grabed by esl phpmod. regards helmut From erkan at speedingtrade.com Wed Jun 23 06:29:18 2010 From: erkan at speedingtrade.com (=?iso-8859-1?B?RXJrYW4g3G5s/A==?=) Date: Wed, 23 Jun 2010 16:29:18 +0300 Subject: [Freeswitch-users] G729 on windows??? Message-ID: <81C2CEF80046FB4F863A60D4347DD33A105B39@server1.st.local> Hi all, can i install the G729 codec and license on the Windows version of Freeswitch? kind regards Erkan Unlu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/deeafeb3/attachment.html From brian at freeswitch.org Wed Jun 23 06:39:49 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Jun 2010 08:39:49 -0500 Subject: [Freeswitch-users] G729 on windows??? In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A105B39@server1.st.local> References: <81C2CEF80046FB4F863A60D4347DD33A105B39@server1.st.local> Message-ID: <9642110D-AA7F-460D-AEBA-E86F0A06EBF2@freeswitch.org> Not yet. /b On Jun 23, 2010, at 8:29 AM, Erkan ?nl? wrote: > Hi all, > > can i install the G729 codec and license on the Windows version of Freeswitch? > > kind regards > Erkan Unlu From helmut.kuper at ewetel.de Wed Jun 23 08:37:51 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 23 Jun 2010 17:37:51 +0200 Subject: [Freeswitch-users] esl phpmod background_job events not complete In-Reply-To: <4C220B5E.1030802@ewetel.de> References: <4C220B5E.1030802@ewetel.de> Message-ID: <4C222A4F.5040103@ewetel.de> Hi, I found the problem. Here is my patch: Index: src/esl_event.c =================================================================== --- src/esl_event.c (Revision 17782) +++ src/esl_event.c (Arbeitskopie) @@ -548,7 +548,7 @@ if (blen && !clen) { snprintf(buf + len, dlen - len, "Content-Length: %d\n\n%s", (int)strlen(event->body), event->body); } else { - snprintf(buf + len, dlen - len, "\n"); + snprintf(buf + len, dlen - len, "\n%s", event->body); } } else { snprintf(buf + len, dlen - len, "\n"); Please review this to avoid side effects. regards Helmut On 23.06.2010 15:25, Helmut Kuper wrote: > Hello > > I found that when I sent a non existent api command via bgapi. The > corresponding background_job event isn't complete: > From rupa at rupa.com Wed Jun 23 08:48:18 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 23 Jun 2010 10:48:18 -0500 Subject: [Freeswitch-users] esl phpmod background_job events not complete In-Reply-To: <4C222A4F.5040103@ewetel.de> References: <4C220B5E.1030802@ewetel.de> <4C222A4F.5040103@ewetel.de> Message-ID: Please open a jira for this. On Wed, Jun 23, 2010 at 10:37 AM, Helmut Kuper wrote: > Hi, > > I found the problem. Here is my patch: > > > Index: src/esl_event.c > =================================================================== > --- src/esl_event.c (Revision 17782) > +++ src/esl_event.c (Arbeitskopie) > @@ -548,7 +548,7 @@ > if (blen && !clen) { > snprintf(buf + len, dlen - len, "Content-Length: > %d\n\n%s", (int)strlen(event->body), event->body); > } else { > - snprintf(buf + len, dlen - len, "\n"); > + snprintf(buf + len, dlen - len, "\n%s", > event->body); > } > } else { > snprintf(buf + len, dlen - len, "\n"); > > > Please review this to avoid side effects. > > > regards > Helmut > > > > On 23.06.2010 15:25, Helmut Kuper wrote: > > Hello > > > > I found that when I sent a non existent api command via bgapi. The > > corresponding background_job event isn't complete: > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/7c352bea/attachment.html From testeador01 at gmail.com Wed Jun 23 08:51:15 2010 From: testeador01 at gmail.com (Milena) Date: Wed, 23 Jun 2010 10:51:15 -0500 Subject: [Freeswitch-users] Understanding mod_shout "licensing" In-Reply-To: References: Message-ID: Hello Tony, The G729 plugin has a license function. Why wouldn't something like > MP3? What is the benefit to FS of accepting a plugin from anyone that > is operating without proper licensing from a copyright or patent > holder? from the same website (http://mp3licensing.com/help/index.html#4): "However, no license is needed for private, non-commercial activities (e.g., home-entertainment, receiving broadcasts and creating a personal music library), not generating revenue or other consideration of any kind or for entities with associated annual gross revenue less than US$ 100 000.00." Not every use of mp3 requires licensing, so if *we* want to use freeswitch with commercial purposes, they make it possible to use mod_shout but it is up to *us* to make sure we are not going against the laws. Just the same way stores sell you a CD/DVD and it is up to you if you're going to reproduce/redistribute the author's music for non-personal uses without his permission. I know it would be nice if we could have all the information given with just installing FreeSWITCH and i think a "licensing feature" is a nice suggestion but I am not quite sure about the funding needed for such a thing. If you need additional dedicated support you can always email consulting at freeswitch.org and check if they provide such services or find legal advise from an expert in your region. As such, is there a peer review board in FS of any sort to keep > potential plugins out of the project unless they properly comply with > copyright and patent laws? If not, why? > > > It seems like a slippery slope to me. I feel these are legitimate > questions, and was wondering if there was a centralized license > function so licenses for things like G729 and MP3 (or anything else) > could be administered and offer to allow a firm to operate a > legitimate server. > About those other 2 questions I don't know, wait for someone else to reply to that. Have a nice day. -Milena -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/c34ca903/attachment.html From brian at freeswitch.org Wed Jun 23 08:56:49 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Jun 2010 10:56:49 -0500 Subject: [Freeswitch-users] Understanding mod_shout "licensing" In-Reply-To: References: Message-ID: <5BA75F03-411D-46EF-B36A-1C241E4A6EFD@freeswitch.org> Since we don't distribute or build mod_shout in any binary form... its not our job to police the patent's. Thats you the end users responsibility. You build it.. you load it... its on you. We have ZERO interest in any patent and licensing of MP3 in FreeSWITCH as a product. mod_shout uses two external libs that are downloaded and built against when YOU build it. /b On Jun 23, 2010, at 10:51 AM, Milena wrote: > > It seems like a slippery slope to me. I feel these are legitimate > questions, and was wondering if there was a centralized license > function so licenses for things like G729 and MP3 (or anything else) > could be administered and offer to allow a firm to operate a > legitimate server. > > About those other 2 questions I don't know, wait for someone else to reply to that. > > Have a nice day. > -Milena From norm at voicenetwork.ca Wed Jun 23 09:02:24 2010 From: norm at voicenetwork.ca (Norman Tomlins) Date: Wed, 23 Jun 2010 12:02:24 -0400 Subject: [Freeswitch-users] Weekly Conference Call - Starts at 1pm EST Message-ID: Hi All, The Weekly Conference call will be starting within the hours. Today's Speaker: June 23, 2010 - Rupa Schomaker - discusses mod_memcache, mod_cidlookup and changes to mod_lcr You can call 1-919-386-9900 to join the Conference. Talk to you soon ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/e1b652bb/attachment.html From helmut.kuper at ewetel.de Wed Jun 23 09:02:36 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 23 Jun 2010 18:02:36 +0200 Subject: [Freeswitch-users] ESL: Question about BACKGROUND_JOB events Message-ID: <4C22301C.3080405@ewetel.de> Hello, today I found that filtering events for a Job-UUID returned by a bgapi command doesn't work well. What I do is this: $con->events("plain", "CHANNEL_HANGUP"); $con->events("plain", "CHANNEL_ANSWER"); $con->events("plain", "BACKGROUND_JOB"); $con->filter("Unique-ID", $a_uuid); $con->filter("Caller-Unique-ID", $a_uuid); $jobid=$con->bgapi("blahblah", ...)->getHeader("Job-UUID"); $con->filter("Job-UUID", $jobid); Then waiting for events. But no BACKGROUND_JOB event is received. Perhaps the BACKGROUND_JOB event is fired before the filter can be applied. When I execute "$con->filter("Event-Name", "BACKGROUND_JOB");" before "$jobid=..." then I'm able to receive it ... and all other but unwanted BACKGROUND_JOB events as well. Any trick to avoid this? regards Helmut From djbinter at gmail.com Wed Jun 23 09:11:18 2010 From: djbinter at gmail.com (DJB International) Date: Wed, 23 Jun 2010 09:11:18 -0700 Subject: [Freeswitch-users] Dialplan failover question In-Reply-To: References: <201006230820.30746.sos@sokhapkin.dyndns.org> <201006230849.42240.sos@sokhapkin.dyndns.org> Message-ID: Would it mean that there are no advantage/disadvantage between the two? It is just how you prefer to write it. Is my assumption correct? Thank you, Dorn B. On Wed, Jun 23, 2010 at 5:59 AM, David Ponzone wrote: > You're absolutely right, I misread your post. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 23/06/2010 ? 14:49, Sergey Okhapkin a ?crit : > > [] sets variables on B leg, but not on A leg. > > On Wednesday 23 June 2010, David Ponzone wrote: > > Sergey, > > > I think you can also do that with the | (pipe) syntax, using [] > > instead of {}. > > > Personnaly, I would say syntax 1 is far more readable. > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > > ? l'intention exclusive de ses destinataires. Toute utilisation ou > > diffusion non autoris?e est interdite. Tout message ?lectronique est > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si > > vous n'?tes pas destinataire de ce message, merci de le d?truire > > imm?diatement et d'avertir l'exp?diteur. > > > Le 23/06/2010 ? 14:20, Sergey Okhapkin a ?crit : > > I prefer the first way, it allows to set leg A channel variables to > > different > > values depending on which gateway completed the call. > > > On Wednesday 23 June 2010, DJB International wrote: > > It's always confused me. I wonder whether there are any differences > > between these 2 dialplans. > > > > > > > > > > > > > > > > > > > > -and- > > > > > > > > > > > > data="sofia/gateway/carrier1/$1|sofia/gateway/carrier2/$1"/> > > > > > > > What is the best way to use it? > > > Thank you, > > Dorn B. > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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/20100623/ce6b2422/attachment-0001.html From david.ponzone at gmail.com Wed Jun 23 09:23:52 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Wed, 23 Jun 2010 18:23:52 +0200 Subject: [Freeswitch-users] Dialplan failover question In-Reply-To: References: <201006230820.30746.sos@sokhapkin.dyndns.org> <201006230849.42240.sos@sokhapkin.dyndns.org> Message-ID: <68EF6B7E-B40E-48FF-B8A0-21F565B80954@gmail.com> Well, as Sergey said, only the first one allows to set variables on leg A. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/06/2010 ? 18:11, DJB International a ?crit : > Would it mean that there are no advantage/disadvantage between the > two? It is just how you prefer to write it. Is my assumption > correct? > > Thank you, > Dorn B. > > On Wed, Jun 23, 2010 at 5:59 AM, David Ponzone > wrote: > You're absolutely right, I misread your post. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 23/06/2010 ? 14:49, Sergey Okhapkin a ?crit : > >> [] sets variables on B leg, but not on A leg. >> >> On Wednesday 23 June 2010, David Ponzone wrote: >>> Sergey, >>> >>> I think you can also do that with the | (pipe) syntax, using [] >>> instead of {}. >>> >>> Personnaly, I would say syntax 1 is far more readable. >>> >>> David Ponzone Direction Technique >>> email: david.ponzone at ipeva.fr >>> tel: 01 74 03 18 97 >>> gsm: 06 66 98 76 34 >>> >>> Service Client IPeva >>> tel: 0811 46 26 26 >>> www.ipeva.fr - www.ipeva-studio.com >>> >>> Ce message et toutes les pi?ces jointes sont confidentiels et >>> ?tablis >>> ? l'intention exclusive de ses destinataires. Toute utilisation ou >>> diffusion non autoris?e est interdite. Tout message ?lectronique est >>> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au >>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si >>> vous n'?tes pas destinataire de ce message, merci de le d?truire >>> imm?diatement et d'avertir l'exp?diteur. >>> >>> Le 23/06/2010 ? 14:20, Sergey Okhapkin a ?crit : >>>> I prefer the first way, it allows to set leg A channel variables to >>>> different >>>> values depending on which gateway completed the call. >>>> >>>> On Wednesday 23 June 2010, DJB International wrote: >>>>> It's always confused me. I wonder whether there are any >>>>> differences >>>>> between these 2 dialplans. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> -and- >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/gateway/carrier1/$1|sofia/gateway/carrier2/$1"/> >>>>> >>>>> >>>>> >>>>> What is the best way to use it? >>>>> >>>>> Thank you, >>>>> Dorn B. >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/c2b87c0e/attachment.html From sos at sokhapkin.dyndns.org Wed Jun 23 09:24:21 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 23 Jun 2010 12:24:21 -0400 Subject: [Freeswitch-users] Dialplan failover question In-Reply-To: References: Message-ID: <201006231224.21257.sos@sokhapkin.dyndns.org> I think there is a difference. Most likely second gateway will not be called in | syntax if the first one will return early media followed by SIP error. On Wednesday 23 June 2010, DJB International wrote: > Would it mean that there are no advantage/disadvantage between the two? It > is just how you prefer to write it. Is my assumption correct? > > Thank you, > Dorn B. > > On Wed, Jun 23, 2010 at 5:59 AM, David Ponzone wrote: > > You're absolutely right, I misread your post. > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > > l'intention exclusive de ses destinataires. Toute utilisation ou > > diffusion non autoris?e est interdite. Tout message ?lectronique est > > susceptible d'alt?ration. **IPeva** d?cline toute responsabilit? au titre > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > > destinataire de ce message, merci de le d?truire imm?diatement et > > d'avertir l'exp?diteur.* > > * > > * > > > > > > > > Le 23/06/2010 ? 14:49, Sergey Okhapkin a ?crit : > > > > [] sets variables on B leg, but not on A leg. > > > > On Wednesday 23 June 2010, David Ponzone wrote: > > > > Sergey, > > > > > > I think you can also do that with the | (pipe) syntax, using [] > > > > instead of {}. > > > > > > Personnaly, I would say syntax 1 is far more readable. > > > > > > David Ponzone Direction Technique > > > > email: david.ponzone at ipeva.fr > > > > tel: 01 74 03 18 97 > > > > gsm: 06 66 98 76 34 > > > > > > Service Client IPeva > > > > tel: 0811 46 26 26 > > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > > > > ? l'intention exclusive de ses destinataires. Toute utilisation ou > > > > diffusion non autoris?e est interdite. Tout message ?lectronique est > > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > > > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si > > > > vous n'?tes pas destinataire de ce message, merci de le d?truire > > > > imm?diatement et d'avertir l'exp?diteur. > > > > > > Le 23/06/2010 ? 14:20, Sergey Okhapkin a ?crit : > > > > I prefer the first way, it allows to set leg A channel variables to > > > > different > > > > values depending on which gateway completed the call. > > > > > > On Wednesday 23 June 2010, DJB International wrote: > > > > It's always confused me. I wonder whether there are any differences > > > > between these 2 dialplans. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -and- > > > > > > > > > > > > > > > > > > > > > > > > > data="sofia/gateway/carrier1/$1|sofia/gateway/carrier2/$1"/> > > > > > > > > > > > > > > What is the best way to use it? > > > > > > Thank you, > > > > Dorn B. > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.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 > From steveu at coppice.org Wed Jun 23 09:33:05 2010 From: steveu at coppice.org (Steve Underwood) Date: Thu, 24 Jun 2010 00:33:05 +0800 Subject: [Freeswitch-users] Understanding mod_shout "licensing" In-Reply-To: References: Message-ID: <4C223741.4030401@coppice.org> On 06/23/2010 11:51 PM, Milena wrote: > Hello Tony, > > The G729 plugin has a license function. Why wouldn't something like > MP3? What is the benefit to FS of accepting a plugin from anyone that > is operating without proper licensing from a copyright or patent > holder? > > from the same website (http://mp3licensing.com/help/index.html#4): > "However, no license is needed for private, non-commercial activities > (e.g., home-entertainment, receiving broadcasts and creating a > personal music library), not generating revenue or other consideration > of any kind or for entities with associated annual gross revenue less > than US$ 100 000.00." > > Not every use of mp3 requires licensing, so if _we_ want to use > freeswitch with commercial purposes, they make it possible to use > mod_shout but it is up to _us_ to make sure we are not going against > the laws. Just the same way stores sell you a CD/DVD and it is up to > you if you're going to reproduce/redistribute the author's music for > non-personal uses without his permission. > > The part you quoted there relates to content licencing, not hardware/software licencing. The whole issue of content licencing seems vague on that site. What constitutes distribution? Does mailing people their voice mails converted to MP3 count as distribution? It seems that every single copy of hardware and software must be licenced, and the low volume charges are clearly listed, as is the minimum of $15,000 per annum. Steve From ranjtech at gmail.com Wed Jun 23 09:33:26 2010 From: ranjtech at gmail.com (RR) Date: Wed, 23 Jun 2010 12:33:26 -0400 Subject: [Freeswitch-users] Diversion header In-Reply-To: References: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org> <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org> <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org> Message-ID: Hello All, I have a call coming in with a Diversion header attached to it but after trying various DialPlan Variables i.e. rdnis, dialed_user and sip_redirect_contact_user_0, I can't seem to extract the rdnis information from the SIP header to then use to route or do any other operation on. Which variable holds this info? How do I get it/use it in the dialplan? Thanks RR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/874cec57/attachment.html From brian at freeswitch.org Wed Jun 23 09:44:56 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Jun 2010 11:44:56 -0500 Subject: [Freeswitch-users] Diversion header In-Reply-To: References: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org> <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org> <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org> Message-ID: ${rdnis} use the info app to dump it. /b On Jun 23, 2010, at 11:33 AM, RR wrote: > Hello All, > > I have a call coming in with a Diversion header attached to it but after trying various DialPlan Variables i.e. rdnis, dialed_user and sip_redirect_contact_user_0, I can't seem to extract the rdnis information from the SIP header to then use to route or do any other operation on. Which variable holds this info? How do I get it/use it in the dialplan? From msc at freeswitch.org Wed Jun 23 09:50:43 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Jun 2010 09:50:43 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly! Message-ID: Everyone start calling in! We'll be starting in just a few minutes... http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_23 -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/cf7be76c/attachment.html From ranjtech at gmail.com Wed Jun 23 10:15:29 2010 From: ranjtech at gmail.com (RR) Date: Wed, 23 Jun 2010 13:15:29 -0400 Subject: [Freeswitch-users] Diversion header In-Reply-To: References: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org> <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org> <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org> Message-ID: Thanks Brian, maybe it's the version of FS that we're using that has a problem but the rdnis variable doesn't even show in the info app dump. The only place I could get the diversion info was in the sip_h_Diversion variable but that has the entire Diversion header instead of just the rdnis. Will have to capture this using regex and then do anything with it. You think this is because of the version of FreeSWITCH we have or is this a bug somewhere. Sorry we can't test this out by upgrading the FS version as this is the only environment we have and its all production :( Thanks RR On Wed, Jun 23, 2010 at 12:44 PM, Brian West wrote: > ${rdnis} use the info app to dump it. > > /b > > On Jun 23, 2010, at 11:33 AM, RR wrote: > > > Hello All, > > > > I have a call coming in with a Diversion header attached to it but after > trying various DialPlan Variables i.e. rdnis, dialed_user and > sip_redirect_contact_user_0, I can't seem to extract the rdnis information > from the SIP header to then use to route or do any other operation on. Which > variable holds this info? How do I get it/use it in the dialplan? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/51137e6e/attachment.html From brian at freeswitch.org Wed Jun 23 10:25:37 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Jun 2010 12:25:37 -0500 Subject: [Freeswitch-users] Diversion header In-Reply-To: References: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org> <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org> <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org> Message-ID: <37A8764C-9C95-424D-8C75-0627AC1C0809@freeswitch.org> No did you even try using ${rdnis} ? Its not going to show up in a dump as rdnis becuse its special read switch_caller.c /b On Jun 23, 2010, at 12:15 PM, RR wrote: > Thanks Brian, > > maybe it's the version of FS that we're using that has a problem but the rdnis variable doesn't even show in the info app dump. The only place I could get the diversion info was in the sip_h_Diversion variable but that has the entire Diversion header instead of just the rdnis. Will have to capture this using regex and then do anything with it. > > You think this is because of the version of FreeSWITCH we have or is this a bug somewhere. Sorry we can't test this out by upgrading the FS version as this is the only environment we have and its all production :( > > Thanks > RR From ranjtech at gmail.com Wed Jun 23 10:40:53 2010 From: ranjtech at gmail.com (RR) Date: Wed, 23 Jun 2010 13:40:53 -0400 Subject: [Freeswitch-users] Diversion header In-Reply-To: <37A8764C-9C95-424D-8C75-0627AC1C0809@freeswitch.org> References: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org> <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org> <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org> <37A8764C-9C95-424D-8C75-0627AC1C0809@freeswitch.org> Message-ID: yeah that was the first variable I'd tried when I started looking into this but it came out empty. This is what I was seeing in the debug dump Regex (FAIL) [Test_Diversion] rdnis() =~ /^(917xxxxxxx)$/ break=on-false even if the condition failed, it usually shows what's in the variable to then compare with the regex Thanks \RR On Wed, Jun 23, 2010 at 1:25 PM, Brian West wrote: > No did you even try using ${rdnis} ? Its not going to show up in a dump as > rdnis becuse its special read switch_caller.c > > /b > > On Jun 23, 2010, at 12:15 PM, RR wrote: > > > Thanks Brian, > > > > maybe it's the version of FS that we're using that has a problem but the > rdnis variable doesn't even show in the info app dump. The only place I > could get the diversion info was in the sip_h_Diversion variable but that > has the entire Diversion header instead of just the rdnis. Will have to > capture this using regex and then do anything with it. > > > > You think this is because of the version of FreeSWITCH we have or is this > a bug somewhere. Sorry we can't test this out by upgrading the FS version as > this is the only environment we have and its all production :( > > > > Thanks > > RR > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/22f30a48/attachment.html From brian at freeswitch.org Wed Jun 23 10:57:21 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Jun 2010 12:57:21 -0500 Subject: [Freeswitch-users] Diversion header In-Reply-To: References: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org> <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org> <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org> <37A8764C-9C95-424D-8C75-0627AC1C0809@freeswitch.org> Message-ID: <59B3FE66-6FE7-48B4-BE51-2392159F340F@freeswitch.org> are you looking at sip_h_Diversion variable? /b On Jun 23, 2010, at 12:40 PM, RR wrote: > yeah that was the first variable I'd tried when I started looking into this but it came out empty. This is what I was seeing in the debug dump > > Regex (FAIL) [Test_Diversion] rdnis() =~ /^(917xxxxxxx)$/ break=on-false > > even if the condition failed, it usually shows what's in the variable to then compare with the regex > > Thanks > \RR > From ranjtech at gmail.com Wed Jun 23 11:07:13 2010 From: ranjtech at gmail.com (RR) Date: Wed, 23 Jun 2010 14:07:13 -0400 Subject: [Freeswitch-users] Diversion header In-Reply-To: <59B3FE66-6FE7-48B4-BE51-2392159F340F@freeswitch.org> References: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org> <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org> <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org> <37A8764C-9C95-424D-8C75-0627AC1C0809@freeswitch.org> <59B3FE66-6FE7-48B4-BE51-2392159F340F@freeswitch.org> Message-ID: I am now considering the ${rdnis} variable is empty. So am doing something like \RR On Wed, Jun 23, 2010 at 1:57 PM, Brian West wrote: > are you looking at sip_h_Diversion variable? > > /b > > On Jun 23, 2010, at 12:40 PM, RR wrote: > > > yeah that was the first variable I'd tried when I started looking into > this but it came out empty. This is what I was seeing in the debug dump > > > > Regex (FAIL) [Test_Diversion] rdnis() =~ /^(917xxxxxxx)$/ break=on-false > > > > even if the condition failed, it usually shows what's in the variable to > then compare with the regex > > > > Thanks > > \RR > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/8dd4d5dd/attachment.html From anthony.minessale at gmail.com Wed Jun 23 11:17:03 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Jun 2010 13:17:03 -0500 Subject: [Freeswitch-users] ESL: phpmod In-Reply-To: <4C21C549.7070700@ewetel.de> References: <4C20DCA4.7090402@ewetel.de> <201006221214.08312.sos@sokhapkin.dyndns.org> <4C20F662.8020504@ewetel.de> <201006221402.56833.sos@sokhapkin.dyndns.org> <4C21021C.5040309@ewetel.de> <4C21B04C.3030904@ewetel.de> <4C21C549.7070700@ewetel.de> Message-ID: try running the "linger" command. that tells FS to not disconnect when the channel hangs up and instead wait for you to disconnect. On Wed, Jun 23, 2010 at 3:26 AM, Helmut Kuper wrote: > Hi again, > > I added now an inbound esl socket back to FS through which I execute the > originate command. And this way it works. When caller hang up during > originate ringing I get the HANGUP event and I can uuid_kill the > originate session. :) > > hmmm... So bgapi originate seems so stick with the channel's outbound > socket when it is called via outbound socket. > > regards > helmut > > On 23.06.2010 08:57, Helmut Kuper wrote: > > Hi Michael, > > > > On 22.06.2010 20:41, Michael Collins wrote: > >> On Tue, Jun 22, 2010 at 11:34 AM, Helmut Kuper >wrote: > >> > >>> Hi, > >>> > >>> hm well, yes, indeed that works, too. Maybe even better. > >>> > >>> But I still have the problem of a hanging command which doesn't comes > >>> back until timeout, answer, cancel, ... on callee's side. > >>> > >> > >> Which command doesn't come back? I'm assuming you mean the "originate" > >> itself and not the bgapi. I don't see how you can avoid a "hang" while > you > >> wait for the 2nd channel (the callee) to be contacted since you have to > wait > >> for him/her to answer. In any case, this should be quite doable either > with > >> api or with bgapi. The difference with bgapi is that you can do other > stuff > >> in your script (if you need to) while api will wait for the result of > the > >> originate before it returns. > > > > > > What I currently do is this: > > > > 1. caller calls in > > 2. FS calls my deamon via socket app (async full) > > 3. Deamon connects the call via ESLconnection > > 4. executing answer > > 5. executing playback > > 6. subscribing to all events > > 7. originate b-leg via bgapi > > 9. waiting for events (this is a loop using esl->recvEventsTimed(1000)) > > 9.1 replaying caller's playback > > 9.2 checking for caller's Hangup > > 9.3 checking for esl->connected=false > > 9.4 checking for b-leg's result (Answer or deny) > > 9.4.1 bridge both legs > > 9.4.2 release esl-socket > > 9.4.3 done > > > > > > When caller hangs up somewhere after step 7 I got sometimes a HANGUP > > event but more often nothing. Trying to write into the esl->socket then > > seems to block until originate has finished. > > > > I guess FS has torn down the socket half way or so in this state. > > > > > > regards > > Helmut > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > -- 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/20100623/fe56554b/attachment.html From anthony.minessale at gmail.com Wed Jun 23 11:54:30 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Jun 2010 13:54:30 -0500 Subject: [Freeswitch-users] esl phpmod background_job events not complete In-Reply-To: <4C222A4F.5040103@ewetel.de> References: <4C220B5E.1030802@ewetel.de> <4C222A4F.5040103@ewetel.de> Message-ID: this patch was wrong, correct fix is added to latest GIT, please switch to GIT for development and also please report bugs on jira bugs reported on the list are not easy to maintain. On Wed, Jun 23, 2010 at 10:37 AM, Helmut Kuper wrote: > Hi, > > I found the problem. Here is my patch: > > > Index: src/esl_event.c > =================================================================== > --- src/esl_event.c (Revision 17782) > +++ src/esl_event.c (Arbeitskopie) > @@ -548,7 +548,7 @@ > if (blen && !clen) { > snprintf(buf + len, dlen - len, "Content-Length: > %d\n\n%s", (int)strlen(event->body), event->body); > } else { > - snprintf(buf + len, dlen - len, "\n"); > + snprintf(buf + len, dlen - len, "\n%s", > event->body); > } > } else { > snprintf(buf + len, dlen - len, "\n"); > > > Please review this to avoid side effects. > > > regards > Helmut > > > > On 23.06.2010 15:25, Helmut Kuper wrote: > > Hello > > > > I found that when I sent a non existent api command via bgapi. The > > corresponding background_job event isn't complete: > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100623/f2ff5c5c/attachment.html From nazim.aghabayov at gmail.com Wed Jun 23 12:08:07 2010 From: nazim.aghabayov at gmail.com (Nazim Aghabayov) Date: Thu, 24 Jun 2010 00:08:07 +0500 Subject: [Freeswitch-users] xml-curl through proxy In-Reply-To: <4C20EE8D.2070903@gmx.net> References: <4C20EE8D.2070903@gmx.net> Message-ID: <4C225B97.8090502@gmail.com> Why http? Why not tcp proxy like rinetd or tcpproxy, they will not alter xml-curl requests, making translation easier. Regards, Nazim On 06/22/2010 10:10 PM, Peter P GMX wrote: > Hello, > > is there a way to pass xml-curl reuqest through a http proxy? For > debugging puposes I would like to reroute destinct requests to another > destination. > > Best regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Wed Jun 23 15:22:54 2010 From: yudha2008 at gmail.com (Baskar) Date: Wed, 23 Jun 2010 18:22:54 -0400 Subject: [Freeswitch-users] SIP Trunk with Private Static IP Message-ID: *Hi All, I have set a freeswitch which has two ethernet cards, one is connected to local lan and the other is connected with our SIP providers trunk.Below is the config of both the ethernet cards* *eth1 ? SIP Provider IP: 172.16.61.27 MASK: 255.255.255.248 GW: 172.16.61.25 eht0 ? Internal LAN IP: 10.10.10.47 MASK: 255.255.255.248 GW: 10.10.10.3 **Dial plan for this SIP provider * *SIP Trunk settings from provider: Protocol: SIP Prefix: 0109#001 Signaling IP: 10.0.0.2 Authentication: IP based Sip port :5060* *There is no user account and password settings for this provider I am able to dial out to local extension using this setting but am not able to make external calls through SIP trunk. The same settings are working fine with trixbox but it does not work with freeswitch. We have another SIP account from one more SIP provider with a username and password and its working fine with freeswitch.Below is the DIal plan of another provider which works fine.* *Dial Plan from the other provider Can you please help me to troubleshoot this issue. Any pointers would be most appreciated. -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/f269a81f/attachment-0001.html From david.ponzone at gmail.com Wed Jun 23 15:45:36 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 24 Jun 2010 00:45:36 +0200 Subject: [Freeswitch-users] SIP Trunk with Private Static IP In-Reply-To: References: Message-ID: <4A20BB52-5B10-4E7E-B86E-C6B27215DCC3@gmail.com> Baskar, there are many inconsistencies/errors in the config you sent. 1) You can't have 2 default routes on a Unix box (well, you can, but the results will be unpredictable). The usual configuration is to have no default route for the LAN (eth0), or possibly some specific static routes, and a default route to the outside (eth1) 2) 10.10.10.3 is not part of the subnet on your eth0 (10.10.10.40/29), so you can't use that as a default route 3) 10.10.10.47 is the broadcast address of the subnet 10.10.10.40/29, so you can't use that. Perhaps 47 was a typo, that would also explain the issue above. 4) if 10.0.0.2 is the IP of your provider, you should really check that you can reach that with the routes you have (see 1 above). If you are running pure private on this box, I would recommend not setting any default route, but rather setting all the required static routes David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 24/06/2010 ? 00:22, Baskar a ?crit : > Hi All, > > I have set a freeswitch which has two ethernet cards, one is > connected to local lan and the other is connected with our SIP > providers trunk.Below is the config of both the ethernet cards > > eth1 ? SIP Provider IP: 172.16.61.27 MASK: 255.255.255.248 GW: > 172.16.61.25 > eht0 ? Internal LAN IP: 10.10.10.47 MASK: 255.255.255.248 GW: > 10.10.10.3 > > Dial plan for this SIP provider > > > > > > > > > > SIP Trunk settings from provider: > > Protocol: SIP > Prefix: 0109#001 > Signaling IP: 10.0.0.2 > Authentication: IP based > Sip port :5060 > > There is no user account and password settings for this provider > > I am able to dial out to local extension using this setting but am > not able to make external calls through SIP trunk. The same settings > are working fine with trixbox but it does not work with freeswitch. > > We have another SIP account from one more SIP provider with a > username and password and its working fine with freeswitch.Below is > the DIal plan of another provider which works fine. > > > Dial Plan from the other provider > > > > data="effective_caller_id_number=12223334444"/> > > > > > > > Can you please help me to troubleshoot this issue. Any pointers > would be most appreciated. > > -- > Thanks with Regards, > > N.Baskar > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/1a1653ef/attachment.html From mcampbellsmith at gmail.com Wed Jun 23 16:32:21 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 24 Jun 2010 09:32:21 +1000 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> References: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> Message-ID: FS version and codecs are shown below, but my config file are probably quite old. But I guess they should still work? All codecs are loaded, and the call works if late negotiation is set on profile internal. As I wrote above: The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here. I forgot to write that Extension 1000 does not support GSM (I want to force transcoding). Is that why FS is filtering out GSM on the b-leg? freeswitch at internal> version FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500) freeswitch at internal> show codecs type,name,ikey codec,ADPCM (IMA),mod_voipcodecs codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_voipcodecs codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_voipcodecs codec,G.726 16k (AAL2),mod_voipcodecs codec,G.726 24k,mod_voipcodecs codec,G.726 24k (AAL2),mod_voipcodecs codec,G.726 32k,mod_voipcodecs codec,G.726 32k (AAL2),mod_voipcodecs codec,G.726 40k,mod_voipcodecs codec,G.726 40k (AAL2),mod_voipcodecs codec,G.729,mod_com_g729 codec,GSM,mod_voipcodecs codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_voipcodecs 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,Speex,mod_speex 25 total. On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone wrote: > Mark, > I confirm that, as I wrote that wiki page (the early negotiation part) :) > Can you really confirm your FS version ? > The parameter you showed is old. > codec-prefs has been replaced in SIP profiles by: > ?? ? > ?? ? > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 23/06/2010 ? 14:43, Mark Campbell-Smith a ?crit : > > Check this good wiki page for how FS negotiates codecs (early > negotiation default): > http://wiki.freeswitch.org/wiki/Codec_Negotiation > > I have this set in my internal profile: > ??? > > and as stated before vars.xml: > > > > Setting late negotiation works (thanks Sergey), but reading the wiki > page, I see the following sentence, which I interpret that GSM should > still be sent: > When FS calls leg B, the list of codecs in outbound-codec-prefs for > the SIP profile is reorganized by pushing the codec negotiated above > for leg A at the top . If B does not accept any of the codecs, the > calls fails, obviously. > > > > On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano > wrote: > > I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? > > Wouldn't fs only get involved when its media server is referred to? > > If the "other endpoint" will only accept G729, doesn't that mean you > > need to change that endpoint to also accept G711 or also license G729 > > in FS? > > On 6/23/10, Mark Campbell-Smith wrote: > > Test Setup: > > vars.xml: > > ? > > ? > > The call setup is extension 1000 calls extension 1020 > > 1. Extension 1000 calls with preferred codec PCMU. ?PCMU is chosen by > > FS as the A-leg codec > > 2. Extension 1020 only supports GSM codec. ?The call fails with Not > > Acceptable Here. > > FS only offers G729 and PCMU to 1020. ?How do I change the number of > > codecs that are offered to an extension? ?I know I can change the > > order in the codec_prefs, but would prefer FS to offer all three > > codecs to an extension. > > ? ?m=audio 23662 RTP/AVP 0 18 101 13 > > ? ?a=rtpmap:0 PCMU/8000 > > ? ?a=rtpmap:18 G729/8000 > > ? ?a=rtpmap:101 telephone-event/8000 > > ? ?a=fmtp:101 0-16 > > ? ?a=rtpmap:13 CN/8000 > > ? ?a=ptime:30 > > Thanks > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > -- > > Sent from my mobile device > > ====================== > > Tony Graziano, Manager > > Telephone: 434.984.8430 > > sip: tgraziano at voice.myitdepartment.net > > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > > Telephone: 434.984.8426 > > sip: helpdesk at voice.myitdepartment.net > > Fax: 434.984.8427 > > Helpdesk Contract Customers: > > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > > Because 31 Oct = 25 Dec. > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 steveo at nxivm.com Wed Jun 23 17:44:09 2010 From: steveo at nxivm.com (Steve O) Date: Wed, 23 Jun 2010 20:44:09 -0400 Subject: [Freeswitch-users] ESL install error Message-ID: <4C22AA59.8030007@nxivm.com> Hello all, I'm trying to add phpmod-install into my Freeswitch installation on 2 different servers (was failing on the first server, so trying another). Here's a snippet of the error I'm encountering on both servers: g++ -I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o cc1plus: warnings being treated as errors esl_wrap.cpp: In function ?void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1047: error: format not a string literal and no format arguments esl_wrap.cpp: In function ?void _wrap_ESLevent_event_get(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1073: error: format not a string literal and no format arguments esl_wrap.cpp: In function ?void _wrap_ESLevent_serialized_string_set(int, zval*, zval**, zval*, int)?: . . . make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 I searched the list for posts related to the [esl_wrap.o] Error 1, followed all the advice that helped others, but am still having the problem. A couple things about my situation I'm considering that might be unique, and the problem (pretty much guessing at this point), is the OS and architectures I'm running. One server's (the 1st server, referenced above) running FreeBSD 8.0 on a 32 bit processor, and the other server's running Ubuntu 10.04 on a 64 bit processor. Thoughts? What other information can I provide to help troubleshoot? Thanks, Steve From steve at barrettsystems.com Wed Jun 23 21:49:31 2010 From: steve at barrettsystems.com (Steve Butterfield) Date: Wed, 23 Jun 2010 22:49:31 -0600 Subject: [Freeswitch-users] Internal Profile Not Valid Message-ID: I am new to using freeswitch but somehow, overnight my internal sip profile became invalid and freeswitch occupies 90+ % of my memory. I have looked over my install and everything seems to be in order. One thing that stands out during boot up is : 2010-06-23 21:48:15.225851 [DEBUG] sofia.c:1317 Creating agent for internal-ipv6 2010-06-23 21:48:17.766457 [ERR] switch_xml.c:1297 Couldnt open /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such file or directory) 2010-06-23 21:48:17.817992 [ERR] switch_xml.c:1297 Couldnt open /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such file or directory) 2010-06-23 21:48:17.906682 [ERR] switch_xml.c:1297 Couldnt open /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such file or directory) 2010-06-23 21:48:17.967737 [ERR] switch_xml.c:1297 Couldnt open /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such file or directory) 2010-06-23 21:48:18.629030 [DEBUG] sofia.c:1353 Created agent for internal-ipv6 Any help would be appreciated. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100623/ae96c754/attachment-0001.html From steveayre at gmail.com Thu Jun 24 01:59:19 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 24 Jun 2010 09:59:19 +0100 Subject: [Freeswitch-users] Internal Profile Not Valid In-Reply-To: References: Message-ID: I think /default/default/ should just be /default/ On 24 June 2010 05:49, Steve Butterfield wrote: > I am new to using freeswitch but somehow, overnight my internal sip profile > became invalid and freeswitch occupies 90+ % of my memory. I have looked > over my install and everything seems to be in order. One thing that stands > out during boot up is : > > 2010-06-23 21:48:15.225851 [DEBUG] sofia.c:1317 Creating agent for > internal-ipv6 > 2010-06-23 21:48:17.766457 [ERR] switch_xml.c:1297 Couldnt open > /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such > file or directory) > 2010-06-23 21:48:17.817992 [ERR] switch_xml.c:1297 Couldnt open > /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such > file or directory) > 2010-06-23 21:48:17.906682 [ERR] switch_xml.c:1297 Couldnt open > /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such > file or directory) > 2010-06-23 21:48:17.967737 [ERR] switch_xml.c:1297 Couldnt open > /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such > file or directory) > 2010-06-23 21:48:18.629030 [DEBUG] sofia.c:1353 Created agent for > internal-ipv6 > > Any help would be appreciated. > Steve > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/417583b5/attachment.html From vkozak at abisoft.spb.ru Thu Jun 24 04:47:24 2010 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Thu, 24 Jun 2010 15:47:24 +0400 Subject: [Freeswitch-users] HOW SHOW CALLER_ID_NUMBER WITHOUT IP Message-ID: hi everybody. I have one problem. I need to show in phone caller_id_number without IP address or caller_id_number + domain without IP address. I use api originate command with origination_caller_id_name and origination_caller_id_number parameters. bgapi originate {origination_caller_id_name=125 at 123.12.13.14,origination_caller_id_number=123 at 123.12.13.14}[origination_uuid=6daa7b7e-97e4-4790-827d-44ff4f40fd18]sofia/internal/sip:1009 at 172.26.10.65:61802;rinstance=437cf350c3a4546f &park() FS cuts value of origination_caller_id_name parameter (delete specified ip) - it's ok. And FS deletes specified ip from value of origination_caller_id_number parameter and adds FS-IP - it's bad. how else can I show caller info? how can I show caller domain? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/5d785f48/attachment.html From david.ponzone at gmail.com Thu Jun 24 04:57:25 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 24 Jun 2010 13:57:25 +0200 Subject: [Freeswitch-users] HOW SHOW CALLER_ID_NUMBER WITHOUT IP In-Reply-To: References: Message-ID: <635E59FE-61DC-48CB-B44B-31AFF48AB772@gmail.com> Use a SIP PROXY. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 24/06/2010 ? 13:47, Kozak Vladimir a ?crit : > hi everybody. > > > I have one problem. I need to show in phone caller_id_number without > IP address or caller_id_number + domain without IP address. I use > api originate command with origination_caller_id_name and > origination_caller_id_number parameters. > bgapi originate{origination_caller_id_name=125 at 123.12.13.14,origination_caller_id_number=123 at 123.12.13.14 > }[origination_uuid=6daa7b7e-97e4-4790-827d-44ff4f40fd18]sofia/ > internal/sip:1009 at 172.26.10.65:61802;rinstance=437cf350c3a4546f > &park() > > FS cuts value of origination_caller_id_name parameter (delete > specified ip) - it's ok. > And FS deletes specified ip from value of > origination_caller_id_number parameter and adds FS-IP - it's bad. > > how else can I show caller info? > how can I show caller domain? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/92c1b20e/attachment.html From irmatov at gmail.com Thu Jun 24 05:58:44 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Thu, 24 Jun 2010 17:58:44 +0500 Subject: [Freeswitch-users] mod_erlang_event problem Message-ID: Hi! I am trying to use mod_erlang_event with freeswitch 1.0.6 on Red Hat Enterprise Linux Server release 5.4, 64-bit server. Erlang version is R13B04. Calls are sent to my erlang application via: My application is very simple: it just prints all events received from freeswitch. The problem is, that call is being terminated immidiately. As far as I can see, phonebooth:launch is called successfully, it returns a pid of a new process. This new process is still alive after the call is finished, and it does not receive any events from freeswitch (if it would, it would print them to screen). Freeswitch log tells me that erlang_outbound_function exits as soon as it gets new pid: 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:1316 got pid! 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:1446 exit erlang_outbound_function What is wrong? As this is not my first try with erlang and freeswitch (previous were successful) I think there's nothing wrong with configuration, but may be the problem is the box itself, old Red Hat? Here's full log from freeswitch: 2010-06-24 17:46:58.265274 [NOTICE] switch_channel.c:669 New Channel sofia/external/1220139 at 10.1.1.1 [93975342-7f8e-11df-bae2-7b19ef829fd0] 2010-06-24 17:46:58.265274 [DEBUG] sofia.c:4153 Channel sofia/external/1220139 at 10.1.1.1 entering state [received][100] 2010-06-24 17:46:58.265274 [DEBUG] sofia.c:4164 Remote SDP: v=0 o=HuaweiSoftX3000 282382 282382 IN IP4 10.1.1.1 s=Sip Call c=IN IP4 172.16.12.1 t=0 0 m=audio 50000 RTP/AVP 8 0 18 4 2 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 2010-06-24 17:46:58.265274 [DEBUG] sofia.c:4273 (sofia/external/1220139 at 10.1.1.1) State Change CS_NEW -> CS_INIT 2010-06-24 17:46:58.265274 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139 at 10.1.1.1 [BREAK] 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1220139 at 10.1.1.1) Running State Change CS_INIT 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:338 (sofia/external/1220139 at 10.1.1.1) State INIT 2010-06-24 17:46:58.265274 [DEBUG] mod_sofia.c:83 sofia/external/1220139 at 10.1.1.1 SOFIA INIT 2010-06-24 17:46:58.265274 [DEBUG] mod_sofia.c:117 (sofia/external/1220139 at 10.1.1.1) State Change CS_INIT -> CS_ROUTING 2010-06-24 17:46:58.265274 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139 at 10.1.1.1 [BREAK] 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:338 (sofia/external/1220139 at 10.1.1.1) State INIT going to sleep 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1220139 at 10.1.1.1) Running State Change CS_ROUTING 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:341 (sofia/external/1220139 at 10.1.1.1) State ROUTING 2010-06-24 17:46:58.265274 [DEBUG] mod_sofia.c:140 sofia/external/1220139 at 10.1.1.1 SOFIA ROUTING 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:77 sofia/external/1220139 at 10.1.1.1 Standard ROUTING 2010-06-24 17:46:58.265274 [INFO] mod_dialplan_xml.c:418 Processing 1220139->0003707057 in context public Dialplan: sofia/external/1220139 at 10.1.1.1 parsing [public->unloop] continue=false Dialplan: sofia/external/1220139 at 10.1.1.1 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/1220139 at 10.1.1.1 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/1220139 at 10.1.1.1 parsing [public->outside_call] continue=true Dialplan: sofia/external/1220139 at 10.1.1.1 Absolute Condition [outside_call] Dialplan: sofia/external/1220139 at 10.1.1.1 Action set(outside_call=true) Dialplan: sofia/external/1220139 at 10.1.1.1 parsing [public->call_debug] continue=true Dialplan: sofia/external/1220139 at 10.1.1.1 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/1220139 at 10.1.1.1 parsing [public->public_extensions] continue=false Dialplan: sofia/external/1220139 at 10.1.1.1 Regex (FAIL) [public_extensions] destination_number(0003707057) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/1220139 at 10.1.1.1 parsing [public->public_did] continue=false Dialplan: sofia/external/1220139 at 10.1.1.1 Regex (FAIL) [public_did] destination_number(0003707057) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/1220139 at 10.1.1.1 parsing [public->phonebooth] continue=false Dialplan: sofia/external/1220139 at 10.1.1.1 Regex (PASS) [phonebooth] destination_number(0003707057) =~ /^000([0-9]+)$/ break=on-false Dialplan: sofia/external/1220139 at 10.1.1.1 Action erlang(phonebooth:launch test at test-server) 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:119 (sofia/external/1220139 at 10.1.1.1) State Change CS_ROUTING -> CS_EXECUTE 2010-06-24 17:46:58.265274 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139 at 10.1.1.1 [BREAK] 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:341 (sofia/external/1220139 at 10.1.1.1) State ROUTING going to sleep 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1220139 at 10.1.1.1) Running State Change CS_EXECUTE 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:348 (sofia/external/1220139 at 10.1.1.1) State EXECUTE 2010-06-24 17:46:58.265274 [DEBUG] mod_sofia.c:226 sofia/external/1220139 at 10.1.1.1 SOFIA EXECUTE 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:157 sofia/external/1220139 at 10.1.1.1 Standard EXECUTE EXECUTE sofia/external/1220139 at 10.1.1.1 set(outside_call=true) 2010-06-24 17:46:58.265274 [DEBUG] mod_dptools.c:816 sofia/external/1220139 at 10.1.1.1 SET [outside_call]=[true] EXECUTE sofia/external/1220139 at 10.1.1.1 erlang(phonebooth:launch test at test-server) 2010-06-24 17:46:58.265274 [DEBUG] mod_erlang_event.c:1403 enter erlang_outbound_function phonebooth:launch test at test-server 2010-06-24 17:46:58.265274 [DEBUG] mod_erlang_event.c:1409 Creating new listener for session 2010-06-24 17:46:58.269275 [DEBUG] mod_erlang_event.c:1418 Launching new listener 2010-06-24 17:46:58.269275 [DEBUG] mod_erlang_event.c:1424 Creating new spawned session for listener 2010-06-24 17:46:58.271280 [DEBUG] mod_erlang_event.c:961 Connection Open 2010-06-24 17:46:58.271280 [DEBUG] mod_erlang_event.c:1291 rpc call: phonebooth:launch(Ref) 2010-06-24 17:46:58.271280 [DEBUG] handle_msg.c:776 Hashed ref to 4.0.0 at freeswitch@test-server 2010-06-24 17:46:58.271280 [DEBUG] handle_msg.c:787 Found waiting slot for 4.0.0 at freeswitch@test-server 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:1316 got pid! 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:1446 exit erlang_outbound_function 2010-06-24 17:46:58.281273 [NOTICE] switch_core_state_machine.c:185 sofia/external/1220139 at 10.1.1.1 has executed the last dialplan instruction, hanging up. 2010-06-24 17:46:58.281273 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/external/1220139 at 10.1.1.1 [CS_EXECUTE] [NORMAL_CLEARING] 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_EXECUTE_COMPLETE to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.281273 [DEBUG] switch_channel.c:2102 Send signal sofia/external/1220139 at 10.1.1.1 [KILL] 2010-06-24 17:46:58.281273 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139 at 10.1.1.1 [BREAK] 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_HANGUP to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:348 (sofia/external/1220139 at 10.1.1.1) State EXECUTE going to sleep 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1220139 at 10.1.1.1) Running State Change CS_HANGUP 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_STATE to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:499 (sofia/external/1220139 at 10.1.1.1) State HANGUP 2010-06-24 17:46:58.281273 [DEBUG] mod_sofia.c:414 Channel sofia/external/1220139 at 10.1.1.1 hanging up, cause: NORMAL_CLEARING 2010-06-24 17:46:58.281273 [DEBUG] mod_sofia.c:476 Responding to INVITE with: 480 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:46 sofia/external/1220139 at 10.1.1.1 Standard HANGUP, cause: NORMAL_CLEARING 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:499 (sofia/external/1220139 at 10.1.1.1) State HANGUP going to sleep 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:333 (sofia/external/1220139 at 10.1.1.1) State Change CS_HANGUP -> CS_REPORTING 2010-06-24 17:46:58.281273 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139 at 10.1.1.1 [BREAK] 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_HANGUP_COMPLETE to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1220139 at 10.1.1.1) Running State Change CS_REPORTING 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:590 (sofia/external/1220139 at 10.1.1.1) State REPORTING 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:53 sofia/external/1220139 at 10.1.1.1 Standard REPORTING, cause: NORMAL_CLEARING 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:590 (sofia/external/1220139 at 10.1.1.1) State REPORTING going to sleep 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:327 (sofia/external/1220139 at 10.1.1.1) State Change CS_REPORTING -> CS_DESTROY 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_STATE to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.281273 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139 at 10.1.1.1 [BREAK] 2010-06-24 17:46:58.281273 [DEBUG] switch_core_session.c:1164 Session 4 (sofia/external/1220139 at 10.1.1.1) Locked, Waiting on external entities 2010-06-24 17:46:58.283274 [NOTICE] switch_core_session.c:1182 Session 4 (sofia/external/1220139 at 10.1.1.1) Ended 2010-06-24 17:46:58.283274 [NOTICE] switch_core_session.c:1184 Close Channel sofia/external/1220139 at 10.1.1.1 [CS_DESTROY] 2010-06-24 17:46:58.283274 [DEBUG] switch_core_state_machine.c:428 (sofia/external/1220139 at 10.1.1.1) Running State Change CS_DESTROY 2010-06-24 17:46:58.283274 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_DESTROY to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.283274 [DEBUG] switch_core_state_machine.c:439 (sofia/external/1220139 at 10.1.1.1) State DESTROY 2010-06-24 17:46:58.283274 [DEBUG] mod_sofia.c:341 sofia/external/1220139 at 10.1.1.1 SOFIA DESTROY 2010-06-24 17:46:58.283274 [DEBUG] switch_core_state_machine.c:60 sofia/external/1220139 at 10.1.1.1 Standard DESTROY 2010-06-24 17:46:58.283274 [DEBUG] switch_core_state_machine.c:439 (sofia/external/1220139 at 10.1.1.1) State DESTROY going to sleep 2010-06-24 17:46:58.283274 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_STATE to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.371269 [WARNING] mod_erlang_event.c:531 Can't locate session 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.371269 [DEBUG] mod_erlang_event.c:586 Notifying new session failed 2010-06-24 17:46:58.371269 [DEBUG] mod_erlang_event.c:323 Removing session element for 93975342-7f8e-11df-bae2-7b19ef829fd0 -- Timur Irmatov, xmpp:irmatov at jabber.ru From mcampbellsmith at gmail.com Thu Jun 24 06:10:13 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 24 Jun 2010 23:10:13 +1000 Subject: [Freeswitch-users] destination ip address Message-ID: Hi Is there a way to get the destination ip address of a registered ATA? I want to be able to set bypass_media based on the ip addresses of the two endpoints. ie if both extensions/endpoints have public ip addresses, I would like to turn on bypass_media. Condition field="network_addr" only does source ip address. Is there a way I can get the destination address? From stephen at mymessage.us Thu Jun 24 06:10:47 2010 From: stephen at mymessage.us (Stephen Cattaneo) Date: Thu, 24 Jun 2010 09:10:47 -0400 Subject: [Freeswitch-users] sched_hangup no hangup cause Message-ID: i have "execute_on_answer=sched_hangup +" + calltimeoutsec + " alotted_timeout" on my bleg but after this hangups and i check blegsession.cause it contains NONE, if the bleg party just hangs up i do get normal_clearing. if there is something im doing wrong or if you need more info just let me know. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/22fe01ca/attachment.html From brian at freeswitch.org Thu Jun 24 06:32:09 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 24 Jun 2010 08:32:09 -0500 Subject: [Freeswitch-users] HOW SHOW CALLER_ID_NUMBER WITHOUT IP In-Reply-To: References: Message-ID: <2CA0C040-5D9E-495A-AFE8-A24963542DDD@freeswitch.org> More than likely its your device thats doing this. Polycom does this also if the IP in the to/from/rpid don't match the ip or hostname of the proxy it'll display the IP in the callerid field. Its not us doing it. /b On Jun 24, 2010, at 6:47 AM, Kozak Vladimir wrote: > hi everybody. > > > I have one problem. I need to show in phone caller_id_number without IP address or caller_id_number + domain without IP address. I use api originate command with origination_caller_id_name and origination_caller_id_number parameters. > bgapi originate {origination_caller_id_name=125 at 123.12.13.14,origination_caller_id_number=123 at 123.12.13.14}[origination_uuid=6daa7b7e-97e4-4790-827d-44ff4f40fd18]sofia/internal/sip:1009 at 172.26.10.65:61802;rinstance=437cf350c3a4546f &park() > > FS cuts value of origination_caller_id_name parameter (delete specified ip) - it's ok. > And FS deletes specified ip from value of origination_caller_id_number parameter and adds FS-IP - it's bad. > > how else can I show caller info? > how can I show caller domain? From brian at freeswitch.org Thu Jun 24 06:39:39 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 24 Jun 2010 08:39:39 -0500 Subject: [Freeswitch-users] ESL install error In-Reply-To: <4C22AA59.8030007@nxivm.com> References: <4C22AA59.8030007@nxivm.com> Message-ID: Remove these two flags. PHP headers are NOT clean. /b On Jun 23, 2010, at 7:44 PM, Steve O wrote: > -Wall -Werror From brian at freeswitch.org Thu Jun 24 06:40:55 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 24 Jun 2010 08:40:55 -0500 Subject: [Freeswitch-users] Internal Profile Not Valid In-Reply-To: References: Message-ID: What revision are you running? Have you tried the latest code? Did you happen to follow the bug guidelines and get a gcore? /b On Jun 23, 2010, at 11:49 PM, Steve Butterfield wrote: > I am new to using freeswitch but somehow, overnight my internal sip profile became invalid and freeswitch occupies 90+ % of my memory. I have looked over my install and everything seems to be in order. One thing that stands out during boot up is : From sameer2k3t at gmail.com Thu Jun 24 06:19:03 2010 From: sameer2k3t at gmail.com (Sameer Khan) Date: Thu, 24 Jun 2010 18:19:03 +0500 Subject: [Freeswitch-users] Hi All Message-ID: I am using Centos and have skype client memory leakage problem. Can any one provide me skype client 2.0.72? I cannot find it.. or a cron discussed at wiki page. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/271e340d/attachment.html From david.ponzone at gmail.com Thu Jun 24 06:43:02 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 24 Jun 2010 15:43:02 +0200 Subject: [Freeswitch-users] destination ip address In-Reply-To: References: Message-ID: <14532D96-AE47-4C4A-9096-411E7F0C70F7@gmail.com> Define destination IP address ? The Public IP in front of the device ? Check sofia_contact. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 24/06/2010 ? 15:10, Mark Campbell-Smith a ?crit : > Hi > > Is there a way to get the destination ip address of a registered ATA? > I want to be able to set bypass_media based on the ip addresses of the > two endpoints. ie if both extensions/endpoints have public ip > addresses, I would like to turn on bypass_media. > > Condition field="network_addr" only does source ip address. Is there > a way I can get the destination address? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/00180ae2/attachment.html From a.afzali2003 at gmail.com Thu Jun 24 07:01:09 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Thu, 24 Jun 2010 18:31:09 +0430 Subject: [Freeswitch-users] Unable to get Origination Caller Id Name / Number Works On FIFO Message-ID: Hi FreeSWITCH, I'm working on routing calls from external profile to a FIFO ( RAFQ1 ). Although I've set the origination_caller_id_name & origination_caller_id_number variables in my dialplan, unfortunately the agent receives calls with Queue , fifo+RAFQ1 Ids. My fifo status and logs as follow. BEST, -- afshin freeswitch at internal> fifo list_verbose {execute_on_answer='unset fifo_hangup_check',fifo_hangup_check=' RAFQ1 at 192.168.128.36 ',origination_caller_id_name=Queue,origination_caller_id_number='fifo+RAFQ1'}{fifo_member_wait=nowait}user/1001 freeswitch at internal> 2010-06-24 13:42:14.673471 [NOTICE] switch_channel.c:776 New Channel sofia/external/22808182 at noProvider[4c530c08-7f96-11df-9204-6904a4602528] 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4293 Channel sofia/external/22808182 at noProvider entering state [received][100] 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4304 Remote SDP: v=0 o=- 6 2 IN IP4 192.168.128.31 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.128.31 t=0 0 m=audio 37558 RTP/AVP 107 0 8 18 101 a=rtpmap:107 BV32/16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [BV32:107:16000:20]/[G7221:115:32000:20] 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [BV32:107:16000:20]/[G7221:107:16000:20] 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [BV32:107:16000:20]/[G722:9:8000:20] 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [BV32:107:16000:20]/[PCMU:0:8000:20] 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [BV32:107:16000:20]/[PCMA:8:8000:20] 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [BV32:107:16000:20]/[GSM:3:8000:20] 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20] 2010-06-24 13:42:14.673471 [DEBUG] switch_core_state_machine.c:314 (sofia/external/22808182 at noProvider) Running State Change CS_NEW 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20] 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20] 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:2462 Set Codec sofia/external/22808182 at noProvider PCMU/8000 20 ms 160 samples 2010-06-24 13:42:14.673471 [DEBUG] switch_core_state_machine.c:320 (sofia/external/22808182 at noProvider) State NEW 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf send/recv payload to 101 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4451 (sofia/external/22808182 at noProvider) State Change CS_NEW -> CS_INIT 2010-06-24 13:42:14.673471 [DEBUG] switch_core_session.c:1027 Send signal sofia/external/22808182 at noProvider [BREAK] 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:314 (sofia/external/22808182 at noProvider) Running State Change CS_INIT 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:338 (sofia/external/22808182 at noProvider) State INIT 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:83 sofia/external/22808182 at noProvider SOFIA INIT 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:117 (sofia/external/22808182 at noProvider) State Change CS_INIT -> CS_ROUTING 2010-06-24 13:42:14.674505 [DEBUG] switch_core_session.c:1027 Send signal sofia/external/22808182 at noProvider [BREAK] 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:338 (sofia/external/22808182 at noProvider) State INIT going to sleep 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:314 (sofia/external/22808182 at noProvider) Running State Change CS_ROUTING 2010-06-24 13:42:14.674505 [DEBUG] switch_channel.c:1474 (sofia/external/22808182 at noProvider) Callstate Change DOWN -> RINGING 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:341 (sofia/external/22808182 at noProvider) State ROUTING 2010-06-24 13:42:14.674505 [DEBUG] switch_channel.c:1333 (sofia/external/22808182 at noProvider) Callstate Change RINGING -> ACTIVE 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:140 sofia/external/22808182 at noProvider SOFIA ROUTING 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:77 sofia/external/22808182 at noProvider Standard ROUTING 2010-06-24 13:42:14.674505 [INFO] mod_dialplan_xml.c:331 Processing Afshin Afzali->1880 in context public Dialplan: sofia/external/22808182 at noProvider parsing [public->unloop] continue=false Dialplan: sofia/external/22808182 at noProvider Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/22808182 at noProvider parsing [public->outside_call] continue=true Dialplan: sofia/external/22808182 at noProvider Absolute Condition [outside_call] Dialplan: sofia/external/22808182 at noProvider Action set(outside_call=true) Dialplan: sofia/external/22808182 at noProvider parsing [public->call_debug] continue=true Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/22808182 at noProvider parsing [public->public_extensions] continue=false Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [public_extensions] destination_number(1880) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/22808182 at noProvider parsing [public->public_did] continue=false Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [public_did] destination_number(1880) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/22808182 at noProvider parsing [public->pub1880_did] continue=false Dialplan: sofia/external/22808182 at noProvider Regex (PASS) [pub1880_did] destination_number(1880) =~ /^(1880)$/ break=on-false Dialplan: sofia/external/22808182 at noProvider Action set(domain_name=192.168.128.36) Dialplan: sofia/external/22808182 at noProvider Action set(sound_prefix=/usr/local/freeswitch/sounds/en/us/callie) Dialplan: sofia/external/22808182 at noProvider Action set(fifo_music=local_stream://moh) Dialplan: sofia/external/22808182 at noProvider Action set(origination_caller_id_name=AFSHIN) Dialplan: sofia/external/22808182 at noProvider Action set(origination_caller_id_number=22808182) Dialplan: sofia/external/22808182 at noProvider Action answer() Dialplan: sofia/external/22808182 at noProvider Action sleep(500) Dialplan: sofia/external/22808182 at noProvider Action playback(ivr/ivr-generic_greeting.wav) Dialplan: sofia/external/22808182 at noProvider Action fifo( RAFQ1 at 192.168.128.36 in) 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:119 (sofia/external/22808182 at noProvider) State Change CS_ROUTING -> CS_EXECUTE 2010-06-24 13:42:14.675445 [DEBUG] switch_core_session.c:1027 Send signal sofia/external/22808182 at noProvider [BREAK] 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:341 (sofia/external/22808182 at noProvider) State ROUTING going to sleep 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:314 (sofia/external/22808182 at noProvider) Running State Change CS_EXECUTE 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:348 (sofia/external/22808182 at noProvider) State EXECUTE 2010-06-24 13:42:14.675445 [DEBUG] mod_sofia.c:233 sofia/external/22808182 at noProvider SOFIA EXECUTE 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:157 sofia/external/22808182 at noProvider Standard EXECUTE EXECUTE sofia/external/22808182 at noProvider set(outside_call=true) 2010-06-24 13:42:14.675445 [DEBUG] mod_dptools.c:843 sofia/external/22808182 at noProvider SET [outside_call]=[true] EXECUTE sofia/external/22808182 at noProvider set(domain_name=192.168.128.36) 2010-06-24 13:42:14.675445 [DEBUG] mod_dptools.c:843 sofia/external/22808182 at noProvider SET [domain_name]=[192.168.128.36] EXECUTE sofia/external/22808182 at noProviderset(sound_prefix=/usr/local/freeswitch/sounds/en/us/callie) 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 sofia/external/22808182 at noProvider SET [sound_prefix]=[/usr/local/freeswitch/sounds/en/us/callie] EXECUTE sofia/external/22808182 at noProviderset(fifo_music=local_stream://moh) 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 sofia/external/22808182 at noProvider SET [fifo_music]=[local_stream://moh] EXECUTE sofia/external/22808182 at noProviderset(origination_caller_id_name=AFSHIN) 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 sofia/external/22808182 at noProvider SET [origination_caller_id_name]=[AFSHIN] EXECUTE sofia/external/22808182 at noProviderset(origination_caller_id_number=22808182) 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 sofia/external/22808182 at noProvider SET [origination_caller_id_number]=[22808182] EXECUTE sofia/external/22808182 at noProvider answer() 2010-06-24 13:42:14.676396 [DEBUG] sofia_glue.c:2702 AUDIO RTP [sofia/external/22808182 at noProvider] 192.168.128.36 port 30436 -> 192.168.128.31 port 37558 codec: 0 ms: 20 2010-06-24 13:42:14.676396 [DEBUG] switch_rtp.c:1408 Starting timer [soft] 160 bytes per 20ms 2010-06-24 13:42:14.678369 [DEBUG] sofia_glue.c:2912 Set 2833 dtmf send payload to 101 2010-06-24 13:42:14.678369 [DEBUG] sofia_glue.c:2917 Set 2833 dtmf receive payload to 101 2010-06-24 13:42:14.678369 [DEBUG] mod_sofia.c:667 Local SDP sofia/external/22808182 at noProvider: v=0 o=FreeSWITCH 1277356498 1277356499 IN IP4 192.168.128.36 s=FreeSWITCH c=IN IP4 192.168.128.36 t=0 0 m=audio 30436 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-06-24 13:42:14.678369 [DEBUG] switch_core_session.c:647 Send signal sofia/external/22808182 at noProvider [BREAK] 2010-06-24 13:42:14.678369 [NOTICE] mod_dptools.c:746 Channel [sofia/external/22808182 at noProvider] has been answered EXECUTE sofia/external/22808182 at noProvider sleep(500) 2010-06-24 13:42:14.678369 [DEBUG] sofia.c:4293 Channel sofia/external/22808182 at noProvider entering state [completed][200] 2010-06-24 13:42:14.718466 [DEBUG] switch_rtp.c:2512 Correct ip/port confirmed. 2010-06-24 13:42:14.782161 [DEBUG] sofia.c:4293 Channel sofia/external/22808182 at noProvider entering state [ready][200] EXECUTE sofia/external/22808182 at noProviderplayback(ivr/ivr-generic_greeting.wav) 2010-06-24 13:42:15.178406 [DEBUG] switch_ivr_play_say.c:1161 Codec Activated L16 at 8000hz 1 channels 20ms 2010-06-24 13:42:21.018797 [DEBUG] switch_ivr_play_say.c:1468 done playing file EXECUTE sofia/external/22808182 at noProvider fifo(RAFQ1 at 192.168.128.36 in) 2010-06-24 13:42:21.018797 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz 2010-06-24 13:42:21.018797 [DEBUG] switch_ivr_play_say.c:1161 Codec Activated L16 at 8000hz 1 channels 20ms 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable string 0 = [execute_on_answer=unset fifo_hangup_check] 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable string 1 = [fifo_hangup_check=RAFQ1 at 192.168.128.36] 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable string 2 = [origination_caller_id_name=Queue] 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable string 3 = [origination_caller_id_number=fifo+RAFQ1] 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable string 4 = [fifo_member_wait=nowait] 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable string 0 = [presence_id=1001 at 192.168.128.36] 2010-06-24 13:42:21.564836 [NOTICE] switch_channel.c:776 New Channel sofia/internal/sip:1001 at 192.168.128.31:63820[506eaaa4-7f96-11df-9205-6904a4602528] 2010-06-24 13:42:21.566617 [DEBUG] mod_sofia.c:3883 (sofia/internal/ sip:1001 at 192.168.128.31:63820) State Change CS_NEW -> CS_INIT 2010-06-24 13:42:21.566617 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] 2010-06-24 13:42:21.566617 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change CS_INIT 2010-06-24 13:42:21.566617 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:1001 at 192.168.128.31:63820) State INIT 2010-06-24 13:42:21.566617 [DEBUG] mod_sofia.c:83 sofia/internal/ sip:1001 at 192.168.128.31:63820 SOFIA INIT 2010-06-24 13:42:21.567514 [DEBUG] mod_sofia.c:117 (sofia/internal/ sip:1001 at 192.168.128.31:63820) State Change CS_INIT -> CS_ROUTING 2010-06-24 13:42:21.567514 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:1001 at 192.168.128.31:63820) State INIT going to sleep 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change CS_ROUTING 2010-06-24 13:42:21.567514 [DEBUG] sofia.c:4293 Channel sofia/internal/ sip:1001 at 192.168.128.31:63820 entering state [calling][0] 2010-06-24 13:42:21.567514 [DEBUG] switch_channel.c:1474 (sofia/internal/ sip:1001 at 192.168.128.31:63820) Callstate Change DOWN -> RINGING 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:1001 at 192.168.128.31:63820) State ROUTING 2010-06-24 13:42:21.567514 [DEBUG] switch_channel.c:1333 (sofia/internal/ sip:1001 at 192.168.128.31:63820) Callstate Change RINGING -> ACTIVE 2010-06-24 13:42:21.567514 [DEBUG] mod_sofia.c:140 sofia/internal/ sip:1001 at 192.168.128.31:63820 SOFIA ROUTING 2010-06-24 13:42:21.567514 [DEBUG] switch_ivr_originate.c:64 (sofia/internal/sip:1001 at 192.168.128.31:63820) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-06-24 13:42:21.567514 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:1001 at 192.168.128.31:63820) State ROUTING going to sleep 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change CS_CONSUME_MEDIA 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:1001 at 192.168.128.31:63820) State CONSUME_MEDIA 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:1001 at 192.168.128.31:63820) State CONSUME_MEDIA going to sleep 2010-06-24 13:42:21.673796 [INFO] sofia.c:662 sofia/internal/ sip:1001 at 192.168.128.31:63820 Update Callee ID to "1001" <1001> 2010-06-24 13:42:21.675668 [DEBUG] sofia.c:4293 Channel sofia/internal/ sip:1001 at 192.168.128.31:63820 entering state [proceeding][180] 2010-06-24 13:42:21.675668 [NOTICE] sofia.c:4365 Ring-Ready sofia/internal/ sip:1001 at 192.168.128.31:63820! 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4293 Channel sofia/internal/ sip:1001 at 192.168.128.31:63820 entering state [completing][200] 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4304 Remote SDP: v=0 o=- 6 2 IN IP4 192.168.128.31 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.128.31 t=0 0 m=audio 63470 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4293 Channel sofia/internal/ sip:1001 at 192.168.128.31:63820 entering state [ready][200] 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20] 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20] 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20] 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:2462 Set Codec sofia/internal/sip:1001 at 192.168.128.31:63820 PCMU/8000 20 ms 160 samples 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3810 Set 2833 dtmf send payload to 101 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:2702 AUDIO RTP [sofia/internal/sip:1001 at 192.168.128.31:63820] 192.168.128.36 port 30704 -> 192.168.128.31 port 63470 codec: 0 ms: 20 2010-06-24 13:42:22.083576 [DEBUG] switch_rtp.c:1408 Starting timer [soft] 160 bytes per 20ms 2010-06-24 13:42:22.084524 [DEBUG] sofia_glue.c:2912 Set 2833 dtmf send payload to 101 2010-06-24 13:42:22.084524 [DEBUG] sofia_glue.c:2917 Set 2833 dtmf receive payload to 101 2010-06-24 13:42:22.084524 [NOTICE] sofia.c:4851 Channel [sofia/internal/ sip:1001 at 192.168.128.31:63820] has been answered 2010-06-24 13:42:22.084524 [DEBUG] switch_channel.c:2549 sofia/internal/ sip:1001 at 192.168.128.31:63820 execute on answer: unset(fifo_hangup_check) EXECUTE sofia/internal/sip:1001 at 192.168.128.31:63820unset(fifo_hangup_check) 2010-06-24 13:42:22.084524 [DEBUG] mod_dptools.c:951 UNSET [fifo_hangup_check] 2010-06-24 13:42:22.084524 [DEBUG] switch_ivr_originate.c:3271 Originate Resulted in Success: [sofia/internal/sip:1001 at 192.168.128.31:63820] 2010-06-24 13:42:22.085421 [DEBUG] switch_ivr_originate.c:3271 Originate Resulted in Success: [sofia/internal/sip:1001 at 192.168.128.31:63820] 2010-06-24 13:42:22.085421 [DEBUG] mod_fifo.c:530 (sofia/internal/ sip:1001 at 192.168.128.31:63820) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2010-06-24 13:42:22.085421 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change CS_EXECUTE 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/sip:1001 at 192.168.128.31:63820) State EXECUTE 2010-06-24 13:42:22.085421 [DEBUG] mod_sofia.c:233 sofia/internal/ sip:1001 at 192.168.128.31:63820 SOFIA EXECUTE 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:157 sofia/internal/sip:1001 at 192.168.128.31:63820 Standard EXECUTE EXECUTE sofia/internal/sip:1001 at 192.168.128.31:63820 fifo( RAFQ1 at 192.168.128.36 out nowait) 2010-06-24 13:42:22.138560 [DEBUG] switch_rtp.c:2512 Correct ip/port confirmed. 2010-06-24 13:42:22.619153 [DEBUG] switch_ivr_play_say.c:1468 done playing file 2010-06-24 13:42:22.619153 [DEBUG] mod_fifo.c:1097 (sofia/external/22808182 at noProvider) State Change CS_EXECUTE -> CS_HIBERNATE 2010-06-24 13:42:22.619153 [DEBUG] switch_core_session.c:1027 Send signal sofia/external/22808182 at noProvider [BREAK] 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:348 (sofia/external/22808182 at noProvider) State EXECUTE going to sleep 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:314 (sofia/external/22808182 at noProvider) Running State Change CS_HIBERNATE 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:363 (sofia/external/22808182 at noProvider) State HIBERNATE 2010-06-24 13:42:22.619153 [DEBUG] mod_sofia.c:214 sofia/external/22808182 at noProvider SOFIA HIBERNATE 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:220 sofia/external/22808182 at noProvider Standard HIBERNATE 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:363 (sofia/external/22808182 at noProvider) State HIBERNATE going to sleep 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:1088 (sofia/external/22808182 at noProvider) State Change CS_HIBERNATE -> CS_CONSUME_MEDIA 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:1027 Send signal sofia/external/22808182 at noProvider [BREAK] 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:314 (sofia/external/22808182 at noProvider) Running State Change CS_CONSUME_MEDIA 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:360 (sofia/external/22808182 at noProvider) State CONSUME_MEDIA 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:647 Send signal sofia/external/22808182 at noProvider [BREAK] 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:665 sofia/external/22808182 at noProvider CUSTOM HOLD 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:647 Send signal sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:360 (sofia/external/22808182 at noProvider) State CONSUME_MEDIA going to sleep 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:1183 (sofia/external/22808182 at noProvider) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:1027 Send signal sofia/external/22808182 at noProvider [BREAK] 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:314 (sofia/external/22808182 at noProvider) Running State Change CS_EXCHANGE_MEDIA 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:351 (sofia/external/22808182 at noProvider) State EXCHANGE_MEDIA 2010-06-24 13:42:22.638406 [DEBUG] mod_sofia.c:538 SOFIA EXCHANGE_MEDIA 2010-06-24 13:42:22.639349 [DEBUG] switch_core_session.c:708 Send signal sofia/external/22808182 at noProvider [BREAK] 2010-06-24 13:42:22.639349 [DEBUG] switch_core_session.c:708 Send signal sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/6b23190a/attachment-0001.html From anthony.minessale at gmail.com Thu Jun 24 07:36:02 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 24 Jun 2010 09:36:02 -0500 Subject: [Freeswitch-users] sched_hangup no hangup cause In-Reply-To: References: Message-ID: Tested this and it works fine on my test box: from the look of your example, you are spelling allotted as alotted (one l instead of 2) which is probably your problem. On Thu, Jun 24, 2010 at 8:10 AM, Stephen Cattaneo wrote: > i have "execute_on_answer=sched_hangup +" + calltimeoutsec + " > alotted_timeout" on my bleg but after this hangups and i check > blegsession.cause it contains NONE, if the bleg party just hangs up i do get > normal_clearing. > > if there is something im doing wrong or if you need more info just let me > know. > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100624/e6227964/attachment.html From anthony.minessale at gmail.com Thu Jun 24 07:41:02 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 24 Jun 2010 09:41:02 -0500 Subject: [Freeswitch-users] Unable to get Origination Caller Id Name / Number Works On FIFO In-Reply-To: References: Message-ID: The outbound calling in fifo does not use the customers caller id because there is no correlation between the caller and the agent until the agent answers and is matched to the nearest caller. if you have 10 customers waiting and 20 agents the module will make 10 outbound calls and pass the agents to the next available caller once they answer not before. If you are using a phone that supports display updates the display will change to the customers caller id once the call is bridged. On Thu, Jun 24, 2010 at 9:01 AM, afshin afzali wrote: > Hi FreeSWITCH, > > I'm working on routing calls from external profile to a FIFO ( RAFQ1 ). > Although I've set the origination_caller_id_name & > origination_caller_id_number variables in my dialplan, unfortunately the > agent receives calls with Queue , fifo+RAFQ1 Ids. My fifo status and logs as > follow. > > BEST, > -- afshin > > > > freeswitch at internal> fifo list_verbose > > waiting_count="0" importance="1"> > > outbound-fail-count="1" next-available="2010-06-24 > 13:49:41">{execute_on_answer='unset fifo_hangup_check',fifo_hangup_check=' > RAFQ1 at 192.168.128.36 > ',origination_caller_id_name=Queue,origination_caller_id_number='fifo+RAFQ1'}{fifo_member_wait=nowait}user/1001 > > > > > caller_count="0" waiting_count="0" importance="0"> > > > > > > > > > > > freeswitch at internal> 2010-06-24 13:42:14.673471 [NOTICE] > switch_channel.c:776 New Channel sofia/external/22808182 at noProvider[4c530c08-7f96-11df-9204-6904a4602528] > 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4293 Channel > sofia/external/22808182 at noProvider entering state [received][100] > 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4304 Remote SDP: > v=0 > o=- 6 2 IN IP4 192.168.128.31 > s=CounterPath eyeBeam 1.5 > c=IN IP4 192.168.128.31 > t=0 0 > m=audio 37558 RTP/AVP 107 0 8 18 101 > a=rtpmap:107 BV32/16000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [BV32:107:16000:20]/[G7221:115:32000:20] > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [BV32:107:16000:20]/[G7221:107:16000:20] > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [BV32:107:16000:20]/[G722:9:8000:20] > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [BV32:107:16000:20]/[PCMU:0:8000:20] > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [BV32:107:16000:20]/[PCMA:8:8000:20] > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [BV32:107:16000:20]/[GSM:3:8000:20] > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:115:32000:20] > 2010-06-24 13:42:14.673471 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/22808182 at noProvider) Running State Change CS_NEW > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:107:16000:20] > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [PCMU:0:8000:20]/[G722:9:8000:20] > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [PCMU:0:8000:20]/[PCMU:0:8000:20] > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:2462 Set Codec > sofia/external/22808182 at noProvider PCMU/8000 20 ms 160 samples > 2010-06-24 13:42:14.673471 [DEBUG] switch_core_state_machine.c:320 > (sofia/external/22808182 at noProvider) State NEW > 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf > send/recv payload to 101 > 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4451 > (sofia/external/22808182 at noProvider) State Change CS_NEW -> CS_INIT > 2010-06-24 13:42:14.673471 [DEBUG] switch_core_session.c:1027 Send signal > sofia/external/22808182 at noProvider [BREAK] > 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/22808182 at noProvider) Running State Change CS_INIT > 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:338 > (sofia/external/22808182 at noProvider) State INIT > 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:83 > sofia/external/22808182 at noProvider SOFIA INIT > 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:117 > (sofia/external/22808182 at noProvider) State Change CS_INIT -> CS_ROUTING > 2010-06-24 13:42:14.674505 [DEBUG] switch_core_session.c:1027 Send signal > sofia/external/22808182 at noProvider [BREAK] > 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:338 > (sofia/external/22808182 at noProvider) State INIT going to sleep > 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/22808182 at noProvider) Running State Change CS_ROUTING > 2010-06-24 13:42:14.674505 [DEBUG] switch_channel.c:1474 > (sofia/external/22808182 at noProvider) Callstate Change DOWN -> RINGING > 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:341 > (sofia/external/22808182 at noProvider) State ROUTING > 2010-06-24 13:42:14.674505 [DEBUG] switch_channel.c:1333 > (sofia/external/22808182 at noProvider) Callstate Change RINGING -> ACTIVE > 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:140 > sofia/external/22808182 at noProvider SOFIA ROUTING > 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:77 > sofia/external/22808182 at noProvider Standard ROUTING > 2010-06-24 13:42:14.674505 [INFO] mod_dialplan_xml.c:331 Processing Afshin > Afzali->1880 in context public > Dialplan: sofia/external/22808182 at noProvider parsing [public->unloop] > continue=false > Dialplan: sofia/external/22808182 at noProvider Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/22808182 at noProvider parsing > [public->outside_call] continue=true > Dialplan: sofia/external/22808182 at noProvider Absolute Condition > [outside_call] > Dialplan: sofia/external/22808182 at noProvider Action set(outside_call=true) > > Dialplan: sofia/external/22808182 at noProvider parsing [public->call_debug] > continue=true > Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [call_debug] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/22808182 at noProvider parsing > [public->public_extensions] continue=false > Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) > [public_extensions] destination_number(1880) =~ /^(10[01][0-9])$/ > break=on-false > Dialplan: sofia/external/22808182 at noProvider parsing [public->public_did] > continue=false > Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [public_did] > destination_number(1880) =~ /^(5551212)$/ break=on-false > Dialplan: sofia/external/22808182 at noProvider parsing [public->pub1880_did] > continue=false > Dialplan: sofia/external/22808182 at noProvider Regex (PASS) [pub1880_did] > destination_number(1880) =~ /^(1880)$/ break=on-false > Dialplan: sofia/external/22808182 at noProvider Action > set(domain_name=192.168.128.36) > Dialplan: sofia/external/22808182 at noProvider Action > set(sound_prefix=/usr/local/freeswitch/sounds/en/us/callie) > Dialplan: sofia/external/22808182 at noProvider Action > set(fifo_music=local_stream://moh) > Dialplan: sofia/external/22808182 at noProvider Action > set(origination_caller_id_name=AFSHIN) > Dialplan: sofia/external/22808182 at noProvider Action > set(origination_caller_id_number=22808182) > Dialplan: sofia/external/22808182 at noProvider Action answer() > Dialplan: sofia/external/22808182 at noProvider Action sleep(500) > Dialplan: sofia/external/22808182 at noProvider Action > playback(ivr/ivr-generic_greeting.wav) > Dialplan: sofia/external/22808182 at noProvider Action fifo( > RAFQ1 at 192.168.128.36 in) > 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:119 > (sofia/external/22808182 at noProvider) State Change CS_ROUTING -> CS_EXECUTE > 2010-06-24 13:42:14.675445 [DEBUG] switch_core_session.c:1027 Send signal > sofia/external/22808182 at noProvider [BREAK] > 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:341 > (sofia/external/22808182 at noProvider) State ROUTING going to sleep > 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/22808182 at noProvider) Running State Change CS_EXECUTE > 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:348 > (sofia/external/22808182 at noProvider) State EXECUTE > 2010-06-24 13:42:14.675445 [DEBUG] mod_sofia.c:233 > sofia/external/22808182 at noProvider SOFIA EXECUTE > 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:157 > sofia/external/22808182 at noProvider Standard EXECUTE > EXECUTE sofia/external/22808182 at noProvider set(outside_call=true) > 2010-06-24 13:42:14.675445 [DEBUG] mod_dptools.c:843 > sofia/external/22808182 at noProvider SET [outside_call]=[true] > EXECUTE sofia/external/22808182 at noProvider set(domain_name=192.168.128.36) > 2010-06-24 13:42:14.675445 [DEBUG] mod_dptools.c:843 > sofia/external/22808182 at noProvider SET [domain_name]=[192.168.128.36] > EXECUTE sofia/external/22808182 at noProviderset(sound_prefix=/usr/local/freeswitch/sounds/en/us/callie) > 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 > sofia/external/22808182 at noProvider SET > [sound_prefix]=[/usr/local/freeswitch/sounds/en/us/callie] > EXECUTE sofia/external/22808182 at noProviderset(fifo_music=local_stream://moh) > 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 > sofia/external/22808182 at noProvider SET [fifo_music]=[local_stream://moh] > EXECUTE sofia/external/22808182 at noProviderset(origination_caller_id_name=AFSHIN) > 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 > sofia/external/22808182 at noProvider SET > [origination_caller_id_name]=[AFSHIN] > EXECUTE sofia/external/22808182 at noProviderset(origination_caller_id_number=22808182) > 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 > sofia/external/22808182 at noProvider SET > [origination_caller_id_number]=[22808182] > EXECUTE sofia/external/22808182 at noProvider answer() > 2010-06-24 13:42:14.676396 [DEBUG] sofia_glue.c:2702 AUDIO RTP > [sofia/external/22808182 at noProvider] 192.168.128.36 port 30436 -> > 192.168.128.31 port 37558 codec: 0 ms: 20 > 2010-06-24 13:42:14.676396 [DEBUG] switch_rtp.c:1408 Starting timer [soft] > 160 bytes per 20ms > 2010-06-24 13:42:14.678369 [DEBUG] sofia_glue.c:2912 Set 2833 dtmf send > payload to 101 > 2010-06-24 13:42:14.678369 [DEBUG] sofia_glue.c:2917 Set 2833 dtmf receive > payload to 101 > 2010-06-24 13:42:14.678369 [DEBUG] mod_sofia.c:667 Local SDP > sofia/external/22808182 at noProvider: > v=0 > o=FreeSWITCH 1277356498 1277356499 IN IP4 192.168.128.36 > s=FreeSWITCH > c=IN IP4 192.168.128.36 > t=0 0 > m=audio 30436 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2010-06-24 13:42:14.678369 [DEBUG] switch_core_session.c:647 Send signal > sofia/external/22808182 at noProvider [BREAK] > 2010-06-24 13:42:14.678369 [NOTICE] mod_dptools.c:746 Channel > [sofia/external/22808182 at noProvider] has been answered > EXECUTE sofia/external/22808182 at noProvider sleep(500) > 2010-06-24 13:42:14.678369 [DEBUG] sofia.c:4293 Channel > sofia/external/22808182 at noProvider entering state [completed][200] > 2010-06-24 13:42:14.718466 [DEBUG] switch_rtp.c:2512 Correct ip/port > confirmed. > 2010-06-24 13:42:14.782161 [DEBUG] sofia.c:4293 Channel > sofia/external/22808182 at noProvider entering state [ready][200] > EXECUTE sofia/external/22808182 at noProviderplayback(ivr/ivr-generic_greeting.wav) > 2010-06-24 13:42:15.178406 [DEBUG] switch_ivr_play_say.c:1161 Codec > Activated L16 at 8000hz 1 channels 20ms > 2010-06-24 13:42:21.018797 [DEBUG] switch_ivr_play_say.c:1468 done playing > file > EXECUTE sofia/external/22808182 at noProvider fifo(RAFQ1 at 192.168.128.36 in) > 2010-06-24 13:42:21.018797 [DEBUG] mod_local_stream.c:421 Opening Stream > [moh/8000] 8000hz > 2010-06-24 13:42:21.018797 [DEBUG] switch_ivr_play_say.c:1161 Codec > Activated L16 at 8000hz 1 channels 20ms > 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable > string 0 = [execute_on_answer=unset fifo_hangup_check] > 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable > string 1 = [fifo_hangup_check=RAFQ1 at 192.168.128.36] > 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable > string 2 = [origination_caller_id_name=Queue] > 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable > string 3 = [origination_caller_id_number=fifo+RAFQ1] > 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable > string 4 = [fifo_member_wait=nowait] > 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable > string 0 = [presence_id=1001 at 192.168.128.36] > 2010-06-24 13:42:21.564836 [NOTICE] switch_channel.c:776 New Channel > sofia/internal/sip:1001 at 192.168.128.31:63820[506eaaa4-7f96-11df-9205-6904a4602528] > 2010-06-24 13:42:21.566617 [DEBUG] mod_sofia.c:3883 (sofia/internal/ > sip:1001 at 192.168.128.31:63820) State Change CS_NEW -> CS_INIT > 2010-06-24 13:42:21.566617 [DEBUG] switch_core_session.c:1027 Send signal > sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] > 2010-06-24 13:42:21.566617 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change > CS_INIT > 2010-06-24 13:42:21.566617 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/sip:1001 at 192.168.128.31:63820) State INIT > 2010-06-24 13:42:21.566617 [DEBUG] mod_sofia.c:83 sofia/internal/ > sip:1001 at 192.168.128.31:63820 SOFIA INIT > 2010-06-24 13:42:21.567514 [DEBUG] mod_sofia.c:117 (sofia/internal/ > sip:1001 at 192.168.128.31:63820) State Change CS_INIT -> CS_ROUTING > 2010-06-24 13:42:21.567514 [DEBUG] switch_core_session.c:1027 Send signal > sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] > 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/sip:1001 at 192.168.128.31:63820) State INIT going to sleep > 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change > CS_ROUTING > 2010-06-24 13:42:21.567514 [DEBUG] sofia.c:4293 Channel sofia/internal/ > sip:1001 at 192.168.128.31:63820 entering state [calling][0] > 2010-06-24 13:42:21.567514 [DEBUG] switch_channel.c:1474 (sofia/internal/ > sip:1001 at 192.168.128.31:63820) Callstate Change DOWN -> RINGING > 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/sip:1001 at 192.168.128.31:63820) State ROUTING > 2010-06-24 13:42:21.567514 [DEBUG] switch_channel.c:1333 (sofia/internal/ > sip:1001 at 192.168.128.31:63820) Callstate Change RINGING -> ACTIVE > 2010-06-24 13:42:21.567514 [DEBUG] mod_sofia.c:140 sofia/internal/ > sip:1001 at 192.168.128.31:63820 SOFIA ROUTING > 2010-06-24 13:42:21.567514 [DEBUG] switch_ivr_originate.c:64 > (sofia/internal/sip:1001 at 192.168.128.31:63820) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2010-06-24 13:42:21.567514 [DEBUG] switch_core_session.c:1027 Send signal > sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] > 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/sip:1001 at 192.168.128.31:63820) State ROUTING going to > sleep > 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change > CS_CONSUME_MEDIA > 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:360 > (sofia/internal/sip:1001 at 192.168.128.31:63820) State CONSUME_MEDIA > 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:360 > (sofia/internal/sip:1001 at 192.168.128.31:63820) State CONSUME_MEDIA going > to sleep > 2010-06-24 13:42:21.673796 [INFO] sofia.c:662 sofia/internal/ > sip:1001 at 192.168.128.31:63820 Update Callee ID to "1001" <1001> > 2010-06-24 13:42:21.675668 [DEBUG] sofia.c:4293 Channel sofia/internal/ > sip:1001 at 192.168.128.31:63820 entering state [proceeding][180] > 2010-06-24 13:42:21.675668 [NOTICE] sofia.c:4365 Ring-Ready sofia/internal/ > sip:1001 at 192.168.128.31:63820! > 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4293 Channel sofia/internal/ > sip:1001 at 192.168.128.31:63820 entering state [completing][200] > 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4304 Remote SDP: > v=0 > o=- 6 2 IN IP4 192.168.128.31 > s=CounterPath X-Lite 3.0 > c=IN IP4 192.168.128.31 > t=0 0 > m=audio 63470 RTP/AVP 0 8 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4293 Channel sofia/internal/ > sip:1001 at 192.168.128.31:63820 entering state [ready][200] > 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:115:32000:20] > 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:107:16000:20] > 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [PCMU:0:8000:20]/[G722:9:8000:20] > 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare > [PCMU:0:8000:20]/[PCMU:0:8000:20] > 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:2462 Set Codec > sofia/internal/sip:1001 at 192.168.128.31:63820 PCMU/8000 20 ms 160 samples > 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3810 Set 2833 dtmf send > payload to 101 > 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:2702 AUDIO RTP > [sofia/internal/sip:1001 at 192.168.128.31:63820] 192.168.128.36 port 30704 > -> 192.168.128.31 port 63470 codec: 0 ms: 20 > 2010-06-24 13:42:22.083576 [DEBUG] switch_rtp.c:1408 Starting timer [soft] > 160 bytes per 20ms > 2010-06-24 13:42:22.084524 [DEBUG] sofia_glue.c:2912 Set 2833 dtmf send > payload to 101 > 2010-06-24 13:42:22.084524 [DEBUG] sofia_glue.c:2917 Set 2833 dtmf receive > payload to 101 > 2010-06-24 13:42:22.084524 [NOTICE] sofia.c:4851 Channel [sofia/internal/ > sip:1001 at 192.168.128.31:63820] has been answered > 2010-06-24 13:42:22.084524 [DEBUG] switch_channel.c:2549 sofia/internal/ > sip:1001 at 192.168.128.31:63820 execute on answer: unset(fifo_hangup_check) > EXECUTE sofia/internal/sip:1001 at 192.168.128.31:63820unset(fifo_hangup_check) > 2010-06-24 13:42:22.084524 [DEBUG] mod_dptools.c:951 UNSET > [fifo_hangup_check] > 2010-06-24 13:42:22.084524 [DEBUG] switch_ivr_originate.c:3271 Originate > Resulted in Success: [sofia/internal/sip:1001 at 192.168.128.31:63820] > 2010-06-24 13:42:22.085421 [DEBUG] switch_ivr_originate.c:3271 Originate > Resulted in Success: [sofia/internal/sip:1001 at 192.168.128.31:63820] > 2010-06-24 13:42:22.085421 [DEBUG] mod_fifo.c:530 (sofia/internal/ > sip:1001 at 192.168.128.31:63820) State Change CS_CONSUME_MEDIA -> CS_EXECUTE > 2010-06-24 13:42:22.085421 [DEBUG] switch_core_session.c:1027 Send signal > sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] > 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change > CS_EXECUTE > 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/sip:1001 at 192.168.128.31:63820) State EXECUTE > 2010-06-24 13:42:22.085421 [DEBUG] mod_sofia.c:233 sofia/internal/ > sip:1001 at 192.168.128.31:63820 SOFIA EXECUTE > 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:157 > sofia/internal/sip:1001 at 192.168.128.31:63820 Standard EXECUTE > EXECUTE sofia/internal/sip:1001 at 192.168.128.31:63820 fifo( > RAFQ1 at 192.168.128.36 out nowait) > 2010-06-24 13:42:22.138560 [DEBUG] switch_rtp.c:2512 Correct ip/port > confirmed. > 2010-06-24 13:42:22.619153 [DEBUG] switch_ivr_play_say.c:1468 done playing > file > 2010-06-24 13:42:22.619153 [DEBUG] mod_fifo.c:1097 > (sofia/external/22808182 at noProvider) State Change CS_EXECUTE -> > CS_HIBERNATE > 2010-06-24 13:42:22.619153 [DEBUG] switch_core_session.c:1027 Send signal > sofia/external/22808182 at noProvider [BREAK] > 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:348 > (sofia/external/22808182 at noProvider) State EXECUTE going to sleep > 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/22808182 at noProvider) Running State Change CS_HIBERNATE > 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:363 > (sofia/external/22808182 at noProvider) State HIBERNATE > 2010-06-24 13:42:22.619153 [DEBUG] mod_sofia.c:214 > sofia/external/22808182 at noProvider SOFIA HIBERNATE > 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:220 > sofia/external/22808182 at noProvider Standard HIBERNATE > 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:363 > (sofia/external/22808182 at noProvider) State HIBERNATE going to sleep > 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:1088 > (sofia/external/22808182 at noProvider) State Change CS_HIBERNATE -> > CS_CONSUME_MEDIA > 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:1027 Send signal > sofia/external/22808182 at noProvider [BREAK] > 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/22808182 at noProvider) Running State Change CS_CONSUME_MEDIA > 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:360 > (sofia/external/22808182 at noProvider) State CONSUME_MEDIA > 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:647 Send signal > sofia/external/22808182 at noProvider [BREAK] > 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:665 > sofia/external/22808182 at noProvider CUSTOM HOLD > 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:647 Send signal > sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] > 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:360 > (sofia/external/22808182 at noProvider) State CONSUME_MEDIA going to sleep > 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:1183 > (sofia/external/22808182 at noProvider) State Change CS_CONSUME_MEDIA -> > CS_EXCHANGE_MEDIA > 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:1027 Send signal > sofia/external/22808182 at noProvider [BREAK] > 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/22808182 at noProvider) Running State Change > CS_EXCHANGE_MEDIA > 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:351 > (sofia/external/22808182 at noProvider) State EXCHANGE_MEDIA > 2010-06-24 13:42:22.638406 [DEBUG] mod_sofia.c:538 SOFIA EXCHANGE_MEDIA > 2010-06-24 13:42:22.639349 [DEBUG] switch_core_session.c:708 Send signal > sofia/external/22808182 at noProvider [BREAK] > 2010-06-24 13:42:22.639349 [DEBUG] switch_core_session.c:708 Send signal > sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/e14a81d0/attachment-0001.html From andrew at hijacked.us Thu Jun 24 07:57:50 2010 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 24 Jun 2010 10:57:50 -0400 Subject: [Freeswitch-users] mod_erlang_event problem In-Reply-To: References: Message-ID: <20100624145749.GA17555@hijacked.us> On Thu, Jun 24, 2010 at 05:58:44PM +0500, Timur Irmatov wrote: > Hi! > > I am trying to use mod_erlang_event with freeswitch 1.0.6 on Red Hat > Enterprise Linux Server release 5.4, 64-bit server. Erlang version is > R13B04. Calls are sent to my erlang application via: > > > > > > > > My application is very simple: it just prints all events received from > freeswitch. The problem is, that call is being terminated immidiately. > As far as I can see, phonebooth:launch is called successfully, it > returns a pid of a new process. This new process is still alive after > the call is finished, and it does not receive any events from > freeswitch (if it would, it would print them to screen). Freeswitch > log tells me that erlang_outbound_function exits as soon as it gets > new pid > Yeah, I think this is a bug - Can you show me the code that's running? I think the bug is in the RPC mechanism vs the ! variant, unfortunately I don't have a box with *both* freeswitch and erlang on it handy at the moment - can you write a simple daemon that waits for the get_pid messages and spawns on demand, as described on the wiki, and see if it works that way (that's the way I usually do it). Thanks, Andrew From a.afzali2003 at gmail.com Thu Jun 24 08:29:01 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Thu, 24 Jun 2010 19:59:01 +0430 Subject: [Freeswitch-users] Unable to get Origination Caller Id Name / Number Works On FIFO In-Reply-To: References: Message-ID: I'm using eyeBeam / X-lite softphone. Supporting SIP Update Message is sufficient ? Thers is not any other configuration that I should do to accomplish this? Regards, -- afshin On Thu, Jun 24, 2010 at 7:11 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The outbound calling in fifo does not use the customers caller id because > there is no correlation between the caller and the agent until the agent > answers and is matched to the nearest caller. > > if you have 10 customers waiting and 20 agents the module will make 10 > outbound calls and pass the agents to the next available caller once they > answer not before. If you are using a phone that supports display updates > the display will change to the customers caller id once the call is bridged. > > > > On Thu, Jun 24, 2010 at 9:01 AM, afshin afzali wrote: > >> Hi FreeSWITCH, >> >> I'm working on routing calls from external profile to a FIFO ( RAFQ1 ). >> Although I've set the origination_caller_id_name & >> origination_caller_id_number variables in my dialplan, unfortunately the >> agent receives calls with Queue , fifo+RAFQ1 Ids. My fifo status and logs as >> follow. >> >> BEST, >> -- afshin >> >> >> >> freeswitch at internal> fifo list_verbose >> >> > waiting_count="0" importance="1"> >> >> > outbound-fail-count="1" next-available="2010-06-24 >> 13:49:41">{execute_on_answer='unset fifo_hangup_check',fifo_hangup_check=' >> RAFQ1 at 192.168.128.36 >> ',origination_caller_id_name=Queue,origination_caller_id_number='fifo+RAFQ1'}{fifo_member_wait=nowait}user/1001 >> >> >> >> >> > caller_count="0" waiting_count="0" importance="0"> >> >> >> >> >> >> >> >> >> >> >> freeswitch at internal> 2010-06-24 13:42:14.673471 [NOTICE] >> switch_channel.c:776 New Channel sofia/external/22808182 at noProvider[4c530c08-7f96-11df-9204-6904a4602528] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4293 Channel >> sofia/external/22808182 at noProvider entering state [received][100] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4304 Remote SDP: >> v=0 >> o=- 6 2 IN IP4 192.168.128.31 >> s=CounterPath eyeBeam 1.5 >> c=IN IP4 192.168.128.31 >> t=0 0 >> m=audio 37558 RTP/AVP 107 0 8 18 101 >> a=rtpmap:107 BV32/16000 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=yes >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [BV32:107:16000:20]/[G7221:115:32000:20] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [BV32:107:16000:20]/[G7221:107:16000:20] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [BV32:107:16000:20]/[G722:9:8000:20] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [BV32:107:16000:20]/[PCMU:0:8000:20] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [BV32:107:16000:20]/[PCMA:8:8000:20] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [BV32:107:16000:20]/[GSM:3:8000:20] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [PCMU:0:8000:20]/[G7221:115:32000:20] >> 2010-06-24 13:42:14.673471 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/22808182 at noProvider) Running State Change CS_NEW >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [PCMU:0:8000:20]/[G7221:107:16000:20] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [PCMU:0:8000:20]/[G722:9:8000:20] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [PCMU:0:8000:20]/[PCMU:0:8000:20] >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:2462 Set Codec >> sofia/external/22808182 at noProvider PCMU/8000 20 ms 160 samples >> 2010-06-24 13:42:14.673471 [DEBUG] switch_core_state_machine.c:320 >> (sofia/external/22808182 at noProvider) State NEW >> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf >> send/recv payload to 101 >> 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4451 >> (sofia/external/22808182 at noProvider) State Change CS_NEW -> CS_INIT >> 2010-06-24 13:42:14.673471 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/external/22808182 at noProvider [BREAK] >> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/22808182 at noProvider) Running State Change CS_INIT >> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:338 >> (sofia/external/22808182 at noProvider) State INIT >> 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:83 >> sofia/external/22808182 at noProvider SOFIA INIT >> 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:117 >> (sofia/external/22808182 at noProvider) State Change CS_INIT -> CS_ROUTING >> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/external/22808182 at noProvider [BREAK] >> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:338 >> (sofia/external/22808182 at noProvider) State INIT going to sleep >> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/22808182 at noProvider) Running State Change CS_ROUTING >> 2010-06-24 13:42:14.674505 [DEBUG] switch_channel.c:1474 >> (sofia/external/22808182 at noProvider) Callstate Change DOWN -> RINGING >> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:341 >> (sofia/external/22808182 at noProvider) State ROUTING >> 2010-06-24 13:42:14.674505 [DEBUG] switch_channel.c:1333 >> (sofia/external/22808182 at noProvider) Callstate Change RINGING -> ACTIVE >> 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:140 >> sofia/external/22808182 at noProvider SOFIA ROUTING >> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:77 >> sofia/external/22808182 at noProvider Standard ROUTING >> 2010-06-24 13:42:14.674505 [INFO] mod_dialplan_xml.c:331 Processing Afshin >> Afzali->1880 in context public >> Dialplan: sofia/external/22808182 at noProvider parsing [public->unloop] >> continue=false >> Dialplan: sofia/external/22808182 at noProvider Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/22808182 at noProvider parsing >> [public->outside_call] continue=true >> Dialplan: sofia/external/22808182 at noProvider Absolute Condition >> [outside_call] >> Dialplan: sofia/external/22808182 at noProvider Action >> set(outside_call=true) >> Dialplan: sofia/external/22808182 at noProvider parsing [public->call_debug] >> continue=true >> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [call_debug] >> ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/22808182 at noProvider parsing >> [public->public_extensions] continue=false >> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) >> [public_extensions] destination_number(1880) =~ /^(10[01][0-9])$/ >> break=on-false >> Dialplan: sofia/external/22808182 at noProvider parsing [public->public_did] >> continue=false >> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [public_did] >> destination_number(1880) =~ /^(5551212)$/ break=on-false >> Dialplan: sofia/external/22808182 at noProvider parsing >> [public->pub1880_did] continue=false >> Dialplan: sofia/external/22808182 at noProvider Regex (PASS) [pub1880_did] >> destination_number(1880) =~ /^(1880)$/ break=on-false >> Dialplan: sofia/external/22808182 at noProvider Action >> set(domain_name=192.168.128.36) >> Dialplan: sofia/external/22808182 at noProvider Action >> set(sound_prefix=/usr/local/freeswitch/sounds/en/us/callie) >> Dialplan: sofia/external/22808182 at noProvider Action >> set(fifo_music=local_stream://moh) >> Dialplan: sofia/external/22808182 at noProvider Action >> set(origination_caller_id_name=AFSHIN) >> Dialplan: sofia/external/22808182 at noProvider Action >> set(origination_caller_id_number=22808182) >> Dialplan: sofia/external/22808182 at noProvider Action answer() >> Dialplan: sofia/external/22808182 at noProvider Action sleep(500) >> Dialplan: sofia/external/22808182 at noProvider Action >> playback(ivr/ivr-generic_greeting.wav) >> Dialplan: sofia/external/22808182 at noProvider Action fifo( >> RAFQ1 at 192.168.128.36 in) >> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:119 >> (sofia/external/22808182 at noProvider) State Change CS_ROUTING -> >> CS_EXECUTE >> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/external/22808182 at noProvider [BREAK] >> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:341 >> (sofia/external/22808182 at noProvider) State ROUTING going to sleep >> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/22808182 at noProvider) Running State Change CS_EXECUTE >> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:348 >> (sofia/external/22808182 at noProvider) State EXECUTE >> 2010-06-24 13:42:14.675445 [DEBUG] mod_sofia.c:233 >> sofia/external/22808182 at noProvider SOFIA EXECUTE >> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:157 >> sofia/external/22808182 at noProvider Standard EXECUTE >> EXECUTE sofia/external/22808182 at noProvider set(outside_call=true) >> 2010-06-24 13:42:14.675445 [DEBUG] mod_dptools.c:843 >> sofia/external/22808182 at noProvider SET [outside_call]=[true] >> EXECUTE sofia/external/22808182 at noProviderset(domain_name=192.168.128.36) >> 2010-06-24 13:42:14.675445 [DEBUG] mod_dptools.c:843 >> sofia/external/22808182 at noProvider SET [domain_name]=[192.168.128.36] >> EXECUTE sofia/external/22808182 at noProviderset(sound_prefix=/usr/local/freeswitch/sounds/en/us/callie) >> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >> sofia/external/22808182 at noProvider SET >> [sound_prefix]=[/usr/local/freeswitch/sounds/en/us/callie] >> EXECUTE sofia/external/22808182 at noProviderset(fifo_music=local_stream://moh) >> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >> sofia/external/22808182 at noProvider SET [fifo_music]=[local_stream://moh] >> EXECUTE sofia/external/22808182 at noProviderset(origination_caller_id_name=AFSHIN) >> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >> sofia/external/22808182 at noProvider SET >> [origination_caller_id_name]=[AFSHIN] >> EXECUTE sofia/external/22808182 at noProviderset(origination_caller_id_number=22808182) >> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >> sofia/external/22808182 at noProvider SET >> [origination_caller_id_number]=[22808182] >> EXECUTE sofia/external/22808182 at noProvider answer() >> 2010-06-24 13:42:14.676396 [DEBUG] sofia_glue.c:2702 AUDIO RTP >> [sofia/external/22808182 at noProvider] 192.168.128.36 port 30436 -> >> 192.168.128.31 port 37558 codec: 0 ms: 20 >> 2010-06-24 13:42:14.676396 [DEBUG] switch_rtp.c:1408 Starting timer [soft] >> 160 bytes per 20ms >> 2010-06-24 13:42:14.678369 [DEBUG] sofia_glue.c:2912 Set 2833 dtmf send >> payload to 101 >> 2010-06-24 13:42:14.678369 [DEBUG] sofia_glue.c:2917 Set 2833 dtmf receive >> payload to 101 >> 2010-06-24 13:42:14.678369 [DEBUG] mod_sofia.c:667 Local SDP >> sofia/external/22808182 at noProvider: >> v=0 >> o=FreeSWITCH 1277356498 1277356499 IN IP4 192.168.128.36 >> s=FreeSWITCH >> c=IN IP4 192.168.128.36 >> t=0 0 >> m=audio 30436 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> >> 2010-06-24 13:42:14.678369 [DEBUG] switch_core_session.c:647 Send signal >> sofia/external/22808182 at noProvider [BREAK] >> 2010-06-24 13:42:14.678369 [NOTICE] mod_dptools.c:746 Channel >> [sofia/external/22808182 at noProvider] has been answered >> EXECUTE sofia/external/22808182 at noProvider sleep(500) >> 2010-06-24 13:42:14.678369 [DEBUG] sofia.c:4293 Channel >> sofia/external/22808182 at noProvider entering state [completed][200] >> 2010-06-24 13:42:14.718466 [DEBUG] switch_rtp.c:2512 Correct ip/port >> confirmed. >> 2010-06-24 13:42:14.782161 [DEBUG] sofia.c:4293 Channel >> sofia/external/22808182 at noProvider entering state [ready][200] >> EXECUTE sofia/external/22808182 at noProviderplayback(ivr/ivr-generic_greeting.wav) >> 2010-06-24 13:42:15.178406 [DEBUG] switch_ivr_play_say.c:1161 Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2010-06-24 13:42:21.018797 [DEBUG] switch_ivr_play_say.c:1468 done playing >> file >> EXECUTE sofia/external/22808182 at noProvider fifo(RAFQ1 at 192.168.128.36 in) >> 2010-06-24 13:42:21.018797 [DEBUG] mod_local_stream.c:421 Opening Stream >> [moh/8000] 8000hz >> 2010-06-24 13:42:21.018797 [DEBUG] switch_ivr_play_say.c:1161 Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >> string 0 = [execute_on_answer=unset fifo_hangup_check] >> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >> string 1 = [fifo_hangup_check=RAFQ1 at 192.168.128.36] >> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >> string 2 = [origination_caller_id_name=Queue] >> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >> string 3 = [origination_caller_id_number=fifo+RAFQ1] >> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >> string 4 = [fifo_member_wait=nowait] >> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >> string 0 = [presence_id=1001 at 192.168.128.36] >> 2010-06-24 13:42:21.564836 [NOTICE] switch_channel.c:776 New Channel >> sofia/internal/sip:1001 at 192.168.128.31:63820[506eaaa4-7f96-11df-9205-6904a4602528] >> 2010-06-24 13:42:21.566617 [DEBUG] mod_sofia.c:3883 (sofia/internal/ >> sip:1001 at 192.168.128.31:63820) State Change CS_NEW -> CS_INIT >> 2010-06-24 13:42:21.566617 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >> 2010-06-24 13:42:21.566617 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >> CS_INIT >> 2010-06-24 13:42:21.566617 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) State INIT >> 2010-06-24 13:42:21.566617 [DEBUG] mod_sofia.c:83 sofia/internal/ >> sip:1001 at 192.168.128.31:63820 SOFIA INIT >> 2010-06-24 13:42:21.567514 [DEBUG] mod_sofia.c:117 (sofia/internal/ >> sip:1001 at 192.168.128.31:63820) State Change CS_INIT -> CS_ROUTING >> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) State INIT going to sleep >> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >> CS_ROUTING >> 2010-06-24 13:42:21.567514 [DEBUG] sofia.c:4293 Channel sofia/internal/ >> sip:1001 at 192.168.128.31:63820 entering state [calling][0] >> 2010-06-24 13:42:21.567514 [DEBUG] switch_channel.c:1474 (sofia/internal/ >> sip:1001 at 192.168.128.31:63820) Callstate Change DOWN -> RINGING >> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) State ROUTING >> 2010-06-24 13:42:21.567514 [DEBUG] switch_channel.c:1333 (sofia/internal/ >> sip:1001 at 192.168.128.31:63820) Callstate Change RINGING -> ACTIVE >> 2010-06-24 13:42:21.567514 [DEBUG] mod_sofia.c:140 sofia/internal/ >> sip:1001 at 192.168.128.31:63820 SOFIA ROUTING >> 2010-06-24 13:42:21.567514 [DEBUG] switch_ivr_originate.c:64 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) State Change CS_ROUTING -> >> CS_CONSUME_MEDIA >> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) State ROUTING going to >> sleep >> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >> CS_CONSUME_MEDIA >> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:360 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) State CONSUME_MEDIA >> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:360 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) State CONSUME_MEDIA going >> to sleep >> 2010-06-24 13:42:21.673796 [INFO] sofia.c:662 sofia/internal/ >> sip:1001 at 192.168.128.31:63820 Update Callee ID to "1001" <1001> >> 2010-06-24 13:42:21.675668 [DEBUG] sofia.c:4293 Channel sofia/internal/ >> sip:1001 at 192.168.128.31:63820 entering state [proceeding][180] >> 2010-06-24 13:42:21.675668 [NOTICE] sofia.c:4365 Ring-Ready >> sofia/internal/sip:1001 at 192.168.128.31:63820! >> 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4293 Channel sofia/internal/ >> sip:1001 at 192.168.128.31:63820 entering state [completing][200] >> 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4304 Remote SDP: >> v=0 >> o=- 6 2 IN IP4 192.168.128.31 >> s=CounterPath X-Lite 3.0 >> c=IN IP4 192.168.128.31 >> t=0 0 >> m=audio 63470 RTP/AVP 0 8 101 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4293 Channel sofia/internal/ >> sip:1001 at 192.168.128.31:63820 entering state [ready][200] >> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [PCMU:0:8000:20]/[G7221:115:32000:20] >> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [PCMU:0:8000:20]/[G7221:107:16000:20] >> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [PCMU:0:8000:20]/[G722:9:8000:20] >> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >> [PCMU:0:8000:20]/[PCMU:0:8000:20] >> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:2462 Set Codec >> sofia/internal/sip:1001 at 192.168.128.31:63820 PCMU/8000 20 ms 160 samples >> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3810 Set 2833 dtmf send >> payload to 101 >> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:2702 AUDIO RTP >> [sofia/internal/sip:1001 at 192.168.128.31:63820] 192.168.128.36 port 30704 >> -> 192.168.128.31 port 63470 codec: 0 ms: 20 >> 2010-06-24 13:42:22.083576 [DEBUG] switch_rtp.c:1408 Starting timer [soft] >> 160 bytes per 20ms >> 2010-06-24 13:42:22.084524 [DEBUG] sofia_glue.c:2912 Set 2833 dtmf send >> payload to 101 >> 2010-06-24 13:42:22.084524 [DEBUG] sofia_glue.c:2917 Set 2833 dtmf receive >> payload to 101 >> 2010-06-24 13:42:22.084524 [NOTICE] sofia.c:4851 Channel [sofia/internal/ >> sip:1001 at 192.168.128.31:63820] has been answered >> 2010-06-24 13:42:22.084524 [DEBUG] switch_channel.c:2549 sofia/internal/ >> sip:1001 at 192.168.128.31:63820 execute on answer: unset(fifo_hangup_check) >> EXECUTE sofia/internal/sip:1001 at 192.168.128.31:63820unset(fifo_hangup_check) >> 2010-06-24 13:42:22.084524 [DEBUG] mod_dptools.c:951 UNSET >> [fifo_hangup_check] >> 2010-06-24 13:42:22.084524 [DEBUG] switch_ivr_originate.c:3271 Originate >> Resulted in Success: [sofia/internal/sip:1001 at 192.168.128.31:63820] >> 2010-06-24 13:42:22.085421 [DEBUG] switch_ivr_originate.c:3271 Originate >> Resulted in Success: [sofia/internal/sip:1001 at 192.168.128.31:63820] >> 2010-06-24 13:42:22.085421 [DEBUG] mod_fifo.c:530 (sofia/internal/ >> sip:1001 at 192.168.128.31:63820) State Change CS_CONSUME_MEDIA -> >> CS_EXECUTE >> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >> CS_EXECUTE >> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/sip:1001 at 192.168.128.31:63820) State EXECUTE >> 2010-06-24 13:42:22.085421 [DEBUG] mod_sofia.c:233 sofia/internal/ >> sip:1001 at 192.168.128.31:63820 SOFIA EXECUTE >> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:157 >> sofia/internal/sip:1001 at 192.168.128.31:63820 Standard EXECUTE >> EXECUTE sofia/internal/sip:1001 at 192.168.128.31:63820 fifo( >> RAFQ1 at 192.168.128.36 out nowait) >> 2010-06-24 13:42:22.138560 [DEBUG] switch_rtp.c:2512 Correct ip/port >> confirmed. >> 2010-06-24 13:42:22.619153 [DEBUG] switch_ivr_play_say.c:1468 done playing >> file >> 2010-06-24 13:42:22.619153 [DEBUG] mod_fifo.c:1097 >> (sofia/external/22808182 at noProvider) State Change CS_EXECUTE -> >> CS_HIBERNATE >> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/external/22808182 at noProvider [BREAK] >> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:348 >> (sofia/external/22808182 at noProvider) State EXECUTE going to sleep >> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/22808182 at noProvider) Running State Change CS_HIBERNATE >> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:363 >> (sofia/external/22808182 at noProvider) State HIBERNATE >> 2010-06-24 13:42:22.619153 [DEBUG] mod_sofia.c:214 >> sofia/external/22808182 at noProvider SOFIA HIBERNATE >> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:220 >> sofia/external/22808182 at noProvider Standard HIBERNATE >> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:363 >> (sofia/external/22808182 at noProvider) State HIBERNATE going to sleep >> 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:1088 >> (sofia/external/22808182 at noProvider) State Change CS_HIBERNATE -> >> CS_CONSUME_MEDIA >> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/external/22808182 at noProvider [BREAK] >> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/22808182 at noProvider) Running State Change >> CS_CONSUME_MEDIA >> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:360 >> (sofia/external/22808182 at noProvider) State CONSUME_MEDIA >> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:647 Send signal >> sofia/external/22808182 at noProvider [BREAK] >> 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:665 >> sofia/external/22808182 at noProvider CUSTOM HOLD >> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:647 Send signal >> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:360 >> (sofia/external/22808182 at noProvider) State CONSUME_MEDIA going to sleep >> 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:1183 >> (sofia/external/22808182 at noProvider) State Change CS_CONSUME_MEDIA -> >> CS_EXCHANGE_MEDIA >> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/external/22808182 at noProvider [BREAK] >> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/22808182 at noProvider) Running State Change >> CS_EXCHANGE_MEDIA >> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:351 >> (sofia/external/22808182 at noProvider) State EXCHANGE_MEDIA >> 2010-06-24 13:42:22.638406 [DEBUG] mod_sofia.c:538 SOFIA EXCHANGE_MEDIA >> 2010-06-24 13:42:22.639349 [DEBUG] switch_core_session.c:708 Send signal >> sofia/external/22808182 at noProvider [BREAK] >> 2010-06-24 13:42:22.639349 [DEBUG] switch_core_session.c:708 Send signal >> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/815d09fe/attachment-0001.html From anthony.minessale at gmail.com Thu Jun 24 08:37:48 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 24 Jun 2010 10:37:48 -0500 Subject: [Freeswitch-users] Unable to get Origination Caller Id Name / Number Works On FIFO In-Reply-To: References: Message-ID: x-lite does not support the update. you should ask them to because it's a very useful and easy to implement feature on their part. On Thu, Jun 24, 2010 at 10:29 AM, afshin afzali wrote: > I'm using eyeBeam / X-lite softphone. Supporting SIP Update Message is > sufficient ? Thers is not any other configuration that I should do to > accomplish this? > > Regards, > -- afshin > > > On Thu, Jun 24, 2010 at 7:11 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> The outbound calling in fifo does not use the customers caller id because >> there is no correlation between the caller and the agent until the agent >> answers and is matched to the nearest caller. >> >> if you have 10 customers waiting and 20 agents the module will make 10 >> outbound calls and pass the agents to the next available caller once they >> answer not before. If you are using a phone that supports display updates >> the display will change to the customers caller id once the call is bridged. >> >> >> >> On Thu, Jun 24, 2010 at 9:01 AM, afshin afzali wrote: >> >>> Hi FreeSWITCH, >>> >>> I'm working on routing calls from external profile to a FIFO ( RAFQ1 ). >>> Although I've set the origination_caller_id_name & >>> origination_caller_id_number variables in my dialplan, unfortunately the >>> agent receives calls with Queue , fifo+RAFQ1 Ids. My fifo status and logs as >>> follow. >>> >>> BEST, >>> -- afshin >>> >>> >>> >>> freeswitch at internal> fifo list_verbose >>> >>> >> waiting_count="0" importance="1"> >>> >>> >> outbound-fail-count="1" next-available="2010-06-24 >>> 13:49:41">{execute_on_answer='unset fifo_hangup_check',fifo_hangup_check=' >>> RAFQ1 at 192.168.128.36 >>> ',origination_caller_id_name=Queue,origination_caller_id_number='fifo+RAFQ1'}{fifo_member_wait=nowait}user/1001 >>> >>> >>> >>> >>> >> caller_count="0" waiting_count="0" importance="0"> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> freeswitch at internal> 2010-06-24 13:42:14.673471 [NOTICE] >>> switch_channel.c:776 New Channel sofia/external/22808182 at noProvider[4c530c08-7f96-11df-9204-6904a4602528] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4293 Channel >>> sofia/external/22808182 at noProvider entering state [received][100] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4304 Remote SDP: >>> v=0 >>> o=- 6 2 IN IP4 192.168.128.31 >>> s=CounterPath eyeBeam 1.5 >>> c=IN IP4 192.168.128.31 >>> t=0 0 >>> m=audio 37558 RTP/AVP 107 0 8 18 101 >>> a=rtpmap:107 BV32/16000 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [BV32:107:16000:20]/[G7221:115:32000:20] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [BV32:107:16000:20]/[G7221:107:16000:20] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [BV32:107:16000:20]/[G722:9:8000:20] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [BV32:107:16000:20]/[PCMU:0:8000:20] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [BV32:107:16000:20]/[PCMA:8:8000:20] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [BV32:107:16000:20]/[GSM:3:8000:20] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [PCMU:0:8000:20]/[G7221:115:32000:20] >>> 2010-06-24 13:42:14.673471 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/22808182 at noProvider) Running State Change CS_NEW >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [PCMU:0:8000:20]/[G7221:107:16000:20] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [PCMU:0:8000:20]/[G722:9:8000:20] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [PCMU:0:8000:20]/[PCMU:0:8000:20] >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:2462 Set Codec >>> sofia/external/22808182 at noProvider PCMU/8000 20 ms 160 samples >>> 2010-06-24 13:42:14.673471 [DEBUG] switch_core_state_machine.c:320 >>> (sofia/external/22808182 at noProvider) State NEW >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf >>> send/recv payload to 101 >>> 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4451 >>> (sofia/external/22808182 at noProvider) State Change CS_NEW -> CS_INIT >>> 2010-06-24 13:42:14.673471 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/external/22808182 at noProvider [BREAK] >>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/22808182 at noProvider) Running State Change CS_INIT >>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/external/22808182 at noProvider) State INIT >>> 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:83 >>> sofia/external/22808182 at noProvider SOFIA INIT >>> 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:117 >>> (sofia/external/22808182 at noProvider) State Change CS_INIT -> CS_ROUTING >>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/external/22808182 at noProvider [BREAK] >>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/external/22808182 at noProvider) State INIT going to sleep >>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/22808182 at noProvider) Running State Change CS_ROUTING >>> 2010-06-24 13:42:14.674505 [DEBUG] switch_channel.c:1474 >>> (sofia/external/22808182 at noProvider) Callstate Change DOWN -> RINGING >>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/external/22808182 at noProvider) State ROUTING >>> 2010-06-24 13:42:14.674505 [DEBUG] switch_channel.c:1333 >>> (sofia/external/22808182 at noProvider) Callstate Change RINGING -> ACTIVE >>> 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:140 >>> sofia/external/22808182 at noProvider SOFIA ROUTING >>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:77 >>> sofia/external/22808182 at noProvider Standard ROUTING >>> 2010-06-24 13:42:14.674505 [INFO] mod_dialplan_xml.c:331 Processing >>> Afshin Afzali->1880 in context public >>> Dialplan: sofia/external/22808182 at noProvider parsing [public->unloop] >>> continue=false >>> Dialplan: sofia/external/22808182 at noProvider Regex (PASS) [unloop] >>> ${unroll_loops}(true) =~ /^true$/ break=on-false >>> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [unloop] >>> ${sip_looped_call}() =~ /^true$/ break=on-false >>> Dialplan: sofia/external/22808182 at noProvider parsing >>> [public->outside_call] continue=true >>> Dialplan: sofia/external/22808182 at noProvider Absolute Condition >>> [outside_call] >>> Dialplan: sofia/external/22808182 at noProvider Action >>> set(outside_call=true) >>> Dialplan: sofia/external/22808182 at noProvider parsing >>> [public->call_debug] continue=true >>> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [call_debug] >>> ${call_debug}(false) =~ /^true$/ break=never >>> Dialplan: sofia/external/22808182 at noProvider parsing >>> [public->public_extensions] continue=false >>> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) >>> [public_extensions] destination_number(1880) =~ /^(10[01][0-9])$/ >>> break=on-false >>> Dialplan: sofia/external/22808182 at noProvider parsing >>> [public->public_did] continue=false >>> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [public_did] >>> destination_number(1880) =~ /^(5551212)$/ break=on-false >>> Dialplan: sofia/external/22808182 at noProvider parsing >>> [public->pub1880_did] continue=false >>> Dialplan: sofia/external/22808182 at noProvider Regex (PASS) [pub1880_did] >>> destination_number(1880) =~ /^(1880)$/ break=on-false >>> Dialplan: sofia/external/22808182 at noProvider Action >>> set(domain_name=192.168.128.36) >>> Dialplan: sofia/external/22808182 at noProvider Action >>> set(sound_prefix=/usr/local/freeswitch/sounds/en/us/callie) >>> Dialplan: sofia/external/22808182 at noProvider Action >>> set(fifo_music=local_stream://moh) >>> Dialplan: sofia/external/22808182 at noProvider Action >>> set(origination_caller_id_name=AFSHIN) >>> Dialplan: sofia/external/22808182 at noProvider Action >>> set(origination_caller_id_number=22808182) >>> Dialplan: sofia/external/22808182 at noProvider Action answer() >>> Dialplan: sofia/external/22808182 at noProvider Action sleep(500) >>> Dialplan: sofia/external/22808182 at noProvider Action >>> playback(ivr/ivr-generic_greeting.wav) >>> Dialplan: sofia/external/22808182 at noProvider Action fifo( >>> RAFQ1 at 192.168.128.36 in) >>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:119 >>> (sofia/external/22808182 at noProvider) State Change CS_ROUTING -> >>> CS_EXECUTE >>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/external/22808182 at noProvider [BREAK] >>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/external/22808182 at noProvider) State ROUTING going to sleep >>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/22808182 at noProvider) Running State Change CS_EXECUTE >>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/external/22808182 at noProvider) State EXECUTE >>> 2010-06-24 13:42:14.675445 [DEBUG] mod_sofia.c:233 >>> sofia/external/22808182 at noProvider SOFIA EXECUTE >>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:157 >>> sofia/external/22808182 at noProvider Standard EXECUTE >>> EXECUTE sofia/external/22808182 at noProvider set(outside_call=true) >>> 2010-06-24 13:42:14.675445 [DEBUG] mod_dptools.c:843 >>> sofia/external/22808182 at noProvider SET [outside_call]=[true] >>> EXECUTE sofia/external/22808182 at noProviderset(domain_name=192.168.128.36) >>> 2010-06-24 13:42:14.675445 [DEBUG] mod_dptools.c:843 >>> sofia/external/22808182 at noProvider SET [domain_name]=[192.168.128.36] >>> EXECUTE sofia/external/22808182 at noProviderset(sound_prefix=/usr/local/freeswitch/sounds/en/us/callie) >>> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >>> sofia/external/22808182 at noProvider SET >>> [sound_prefix]=[/usr/local/freeswitch/sounds/en/us/callie] >>> EXECUTE sofia/external/22808182 at noProviderset(fifo_music=local_stream://moh) >>> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >>> sofia/external/22808182 at noProvider SET [fifo_music]=[local_stream://moh] >>> EXECUTE sofia/external/22808182 at noProviderset(origination_caller_id_name=AFSHIN) >>> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >>> sofia/external/22808182 at noProvider SET >>> [origination_caller_id_name]=[AFSHIN] >>> EXECUTE sofia/external/22808182 at noProviderset(origination_caller_id_number=22808182) >>> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >>> sofia/external/22808182 at noProvider SET >>> [origination_caller_id_number]=[22808182] >>> EXECUTE sofia/external/22808182 at noProvider answer() >>> 2010-06-24 13:42:14.676396 [DEBUG] sofia_glue.c:2702 AUDIO RTP >>> [sofia/external/22808182 at noProvider] 192.168.128.36 port 30436 -> >>> 192.168.128.31 port 37558 codec: 0 ms: 20 >>> 2010-06-24 13:42:14.676396 [DEBUG] switch_rtp.c:1408 Starting timer >>> [soft] 160 bytes per 20ms >>> 2010-06-24 13:42:14.678369 [DEBUG] sofia_glue.c:2912 Set 2833 dtmf send >>> payload to 101 >>> 2010-06-24 13:42:14.678369 [DEBUG] sofia_glue.c:2917 Set 2833 dtmf >>> receive payload to 101 >>> 2010-06-24 13:42:14.678369 [DEBUG] mod_sofia.c:667 Local SDP >>> sofia/external/22808182 at noProvider: >>> v=0 >>> o=FreeSWITCH 1277356498 1277356499 IN IP4 192.168.128.36 >>> s=FreeSWITCH >>> c=IN IP4 192.168.128.36 >>> t=0 0 >>> m=audio 30436 RTP/AVP 0 101 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=silenceSupp:off - - - - >>> a=ptime:20 >>> a=sendrecv >>> >>> 2010-06-24 13:42:14.678369 [DEBUG] switch_core_session.c:647 Send signal >>> sofia/external/22808182 at noProvider [BREAK] >>> 2010-06-24 13:42:14.678369 [NOTICE] mod_dptools.c:746 Channel >>> [sofia/external/22808182 at noProvider] has been answered >>> EXECUTE sofia/external/22808182 at noProvider sleep(500) >>> 2010-06-24 13:42:14.678369 [DEBUG] sofia.c:4293 Channel >>> sofia/external/22808182 at noProvider entering state [completed][200] >>> 2010-06-24 13:42:14.718466 [DEBUG] switch_rtp.c:2512 Correct ip/port >>> confirmed. >>> 2010-06-24 13:42:14.782161 [DEBUG] sofia.c:4293 Channel >>> sofia/external/22808182 at noProvider entering state [ready][200] >>> EXECUTE sofia/external/22808182 at noProviderplayback(ivr/ivr-generic_greeting.wav) >>> 2010-06-24 13:42:15.178406 [DEBUG] switch_ivr_play_say.c:1161 Codec >>> Activated L16 at 8000hz 1 channels 20ms >>> 2010-06-24 13:42:21.018797 [DEBUG] switch_ivr_play_say.c:1468 done >>> playing file >>> EXECUTE sofia/external/22808182 at noProvider fifo(RAFQ1 at 192.168.128.36 in) >>> 2010-06-24 13:42:21.018797 [DEBUG] mod_local_stream.c:421 Opening Stream >>> [moh/8000] 8000hz >>> 2010-06-24 13:42:21.018797 [DEBUG] switch_ivr_play_say.c:1161 Codec >>> Activated L16 at 8000hz 1 channels 20ms >>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>> string 0 = [execute_on_answer=unset fifo_hangup_check] >>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>> string 1 = [fifo_hangup_check=RAFQ1 at 192.168.128.36] >>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>> string 2 = [origination_caller_id_name=Queue] >>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>> string 3 = [origination_caller_id_number=fifo+RAFQ1] >>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>> string 4 = [fifo_member_wait=nowait] >>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>> string 0 = [presence_id=1001 at 192.168.128.36] >>> 2010-06-24 13:42:21.564836 [NOTICE] switch_channel.c:776 New Channel >>> sofia/internal/sip:1001 at 192.168.128.31:63820[506eaaa4-7f96-11df-9205-6904a4602528] >>> 2010-06-24 13:42:21.566617 [DEBUG] mod_sofia.c:3883 (sofia/internal/ >>> sip:1001 at 192.168.128.31:63820) State Change CS_NEW -> CS_INIT >>> 2010-06-24 13:42:21.566617 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>> 2010-06-24 13:42:21.566617 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >>> CS_INIT >>> 2010-06-24 13:42:21.566617 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State INIT >>> 2010-06-24 13:42:21.566617 [DEBUG] mod_sofia.c:83 sofia/internal/ >>> sip:1001 at 192.168.128.31:63820 SOFIA INIT >>> 2010-06-24 13:42:21.567514 [DEBUG] mod_sofia.c:117 (sofia/internal/ >>> sip:1001 at 192.168.128.31:63820) State Change CS_INIT -> CS_ROUTING >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State INIT going to sleep >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >>> CS_ROUTING >>> 2010-06-24 13:42:21.567514 [DEBUG] sofia.c:4293 Channel sofia/internal/ >>> sip:1001 at 192.168.128.31:63820 entering state [calling][0] >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_channel.c:1474 (sofia/internal/ >>> sip:1001 at 192.168.128.31:63820) Callstate Change DOWN -> RINGING >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State ROUTING >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_channel.c:1333 (sofia/internal/ >>> sip:1001 at 192.168.128.31:63820) Callstate Change RINGING -> ACTIVE >>> 2010-06-24 13:42:21.567514 [DEBUG] mod_sofia.c:140 sofia/internal/ >>> sip:1001 at 192.168.128.31:63820 SOFIA ROUTING >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_ivr_originate.c:64 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State Change CS_ROUTING >>> -> CS_CONSUME_MEDIA >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State ROUTING going to >>> sleep >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >>> CS_CONSUME_MEDIA >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:360 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State CONSUME_MEDIA >>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:360 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State CONSUME_MEDIA going >>> to sleep >>> 2010-06-24 13:42:21.673796 [INFO] sofia.c:662 sofia/internal/ >>> sip:1001 at 192.168.128.31:63820 Update Callee ID to "1001" <1001> >>> 2010-06-24 13:42:21.675668 [DEBUG] sofia.c:4293 Channel sofia/internal/ >>> sip:1001 at 192.168.128.31:63820 entering state [proceeding][180] >>> 2010-06-24 13:42:21.675668 [NOTICE] sofia.c:4365 Ring-Ready >>> sofia/internal/sip:1001 at 192.168.128.31:63820! >>> 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4293 Channel sofia/internal/ >>> sip:1001 at 192.168.128.31:63820 entering state [completing][200] >>> 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4304 Remote SDP: >>> v=0 >>> o=- 6 2 IN IP4 192.168.128.31 >>> s=CounterPath X-Lite 3.0 >>> c=IN IP4 192.168.128.31 >>> t=0 0 >>> m=audio 63470 RTP/AVP 0 8 101 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> >>> 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4293 Channel sofia/internal/ >>> sip:1001 at 192.168.128.31:63820 entering state [ready][200] >>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [PCMU:0:8000:20]/[G7221:115:32000:20] >>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [PCMU:0:8000:20]/[G7221:107:16000:20] >>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [PCMU:0:8000:20]/[G722:9:8000:20] >>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>> [PCMU:0:8000:20]/[PCMU:0:8000:20] >>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:2462 Set Codec >>> sofia/internal/sip:1001 at 192.168.128.31:63820 PCMU/8000 20 ms 160 samples >>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3810 Set 2833 dtmf send >>> payload to 101 >>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:2702 AUDIO RTP >>> [sofia/internal/sip:1001 at 192.168.128.31:63820] 192.168.128.36 port 30704 >>> -> 192.168.128.31 port 63470 codec: 0 ms: 20 >>> 2010-06-24 13:42:22.083576 [DEBUG] switch_rtp.c:1408 Starting timer >>> [soft] 160 bytes per 20ms >>> 2010-06-24 13:42:22.084524 [DEBUG] sofia_glue.c:2912 Set 2833 dtmf send >>> payload to 101 >>> 2010-06-24 13:42:22.084524 [DEBUG] sofia_glue.c:2917 Set 2833 dtmf >>> receive payload to 101 >>> 2010-06-24 13:42:22.084524 [NOTICE] sofia.c:4851 Channel [sofia/internal/ >>> sip:1001 at 192.168.128.31:63820] has been answered >>> 2010-06-24 13:42:22.084524 [DEBUG] switch_channel.c:2549 sofia/internal/ >>> sip:1001 at 192.168.128.31:63820 execute on answer: >>> unset(fifo_hangup_check) >>> EXECUTE sofia/internal/sip:1001 at 192.168.128.31:63820unset(fifo_hangup_check) >>> 2010-06-24 13:42:22.084524 [DEBUG] mod_dptools.c:951 UNSET >>> [fifo_hangup_check] >>> 2010-06-24 13:42:22.084524 [DEBUG] switch_ivr_originate.c:3271 Originate >>> Resulted in Success: [sofia/internal/sip:1001 at 192.168.128.31:63820] >>> 2010-06-24 13:42:22.085421 [DEBUG] switch_ivr_originate.c:3271 Originate >>> Resulted in Success: [sofia/internal/sip:1001 at 192.168.128.31:63820] >>> 2010-06-24 13:42:22.085421 [DEBUG] mod_fifo.c:530 (sofia/internal/ >>> sip:1001 at 192.168.128.31:63820) State Change CS_CONSUME_MEDIA -> >>> CS_EXECUTE >>> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >>> CS_EXECUTE >>> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State EXECUTE >>> 2010-06-24 13:42:22.085421 [DEBUG] mod_sofia.c:233 sofia/internal/ >>> sip:1001 at 192.168.128.31:63820 SOFIA EXECUTE >>> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:157 >>> sofia/internal/sip:1001 at 192.168.128.31:63820 Standard EXECUTE >>> EXECUTE sofia/internal/sip:1001 at 192.168.128.31:63820 fifo( >>> RAFQ1 at 192.168.128.36 out nowait) >>> 2010-06-24 13:42:22.138560 [DEBUG] switch_rtp.c:2512 Correct ip/port >>> confirmed. >>> 2010-06-24 13:42:22.619153 [DEBUG] switch_ivr_play_say.c:1468 done >>> playing file >>> 2010-06-24 13:42:22.619153 [DEBUG] mod_fifo.c:1097 >>> (sofia/external/22808182 at noProvider) State Change CS_EXECUTE -> >>> CS_HIBERNATE >>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/external/22808182 at noProvider [BREAK] >>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/external/22808182 at noProvider) State EXECUTE going to sleep >>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/22808182 at noProvider) Running State Change CS_HIBERNATE >>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:363 >>> (sofia/external/22808182 at noProvider) State HIBERNATE >>> 2010-06-24 13:42:22.619153 [DEBUG] mod_sofia.c:214 >>> sofia/external/22808182 at noProvider SOFIA HIBERNATE >>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:220 >>> sofia/external/22808182 at noProvider Standard HIBERNATE >>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:363 >>> (sofia/external/22808182 at noProvider) State HIBERNATE going to sleep >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:1088 >>> (sofia/external/22808182 at noProvider) State Change CS_HIBERNATE -> >>> CS_CONSUME_MEDIA >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/external/22808182 at noProvider [BREAK] >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/22808182 at noProvider) Running State Change >>> CS_CONSUME_MEDIA >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:360 >>> (sofia/external/22808182 at noProvider) State CONSUME_MEDIA >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:647 Send signal >>> sofia/external/22808182 at noProvider [BREAK] >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:665 >>> sofia/external/22808182 at noProvider CUSTOM HOLD >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:647 Send signal >>> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:360 >>> (sofia/external/22808182 at noProvider) State CONSUME_MEDIA going to sleep >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:1183 >>> (sofia/external/22808182 at noProvider) State Change CS_CONSUME_MEDIA -> >>> CS_EXCHANGE_MEDIA >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/external/22808182 at noProvider [BREAK] >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/22808182 at noProvider) Running State Change >>> CS_EXCHANGE_MEDIA >>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:351 >>> (sofia/external/22808182 at noProvider) State EXCHANGE_MEDIA >>> 2010-06-24 13:42:22.638406 [DEBUG] mod_sofia.c:538 SOFIA EXCHANGE_MEDIA >>> 2010-06-24 13:42:22.639349 [DEBUG] switch_core_session.c:708 Send signal >>> sofia/external/22808182 at noProvider [BREAK] >>> 2010-06-24 13:42:22.639349 [DEBUG] switch_core_session.c:708 Send signal >>> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>> >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100624/bc4face9/attachment-0001.html From a.afzali2003 at gmail.com Thu Jun 24 08:52:53 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Thu, 24 Jun 2010 20:22:53 +0430 Subject: [Freeswitch-users] Unable to get Origination Caller Id Name / Number Works On FIFO In-Reply-To: References: Message-ID: Thank you Anthony very much. Best On Thu, Jun 24, 2010 at 8:07 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > x-lite does not support the update. > you should ask them to because it's a very useful and easy to implement > feature on their part. > > > > On Thu, Jun 24, 2010 at 10:29 AM, afshin afzali wrote: > >> I'm using eyeBeam / X-lite softphone. Supporting SIP Update Message is >> sufficient ? Thers is not any other configuration that I should do to >> accomplish this? >> >> Regards, >> -- afshin >> >> >> On Thu, Jun 24, 2010 at 7:11 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> The outbound calling in fifo does not use the customers caller id because >>> there is no correlation between the caller and the agent until the agent >>> answers and is matched to the nearest caller. >>> >>> if you have 10 customers waiting and 20 agents the module will make 10 >>> outbound calls and pass the agents to the next available caller once they >>> answer not before. If you are using a phone that supports display updates >>> the display will change to the customers caller id once the call is bridged. >>> >>> >>> >>> On Thu, Jun 24, 2010 at 9:01 AM, afshin afzali wrote: >>> >>>> Hi FreeSWITCH, >>>> >>>> I'm working on routing calls from external profile to a FIFO ( RAFQ1 ). >>>> Although I've set the origination_caller_id_name & >>>> origination_caller_id_number variables in my dialplan, unfortunately the >>>> agent receives calls with Queue , fifo+RAFQ1 Ids. My fifo status and logs as >>>> follow. >>>> >>>> BEST, >>>> -- afshin >>>> >>>> >>>> >>>> freeswitch at internal> fifo list_verbose >>>> >>>> >>> waiting_count="0" importance="1"> >>>> >>>> >>> outbound-fail-count="1" next-available="2010-06-24 >>>> 13:49:41">{execute_on_answer='unset fifo_hangup_check',fifo_hangup_check=' >>>> RAFQ1 at 192.168.128.36 >>>> ',origination_caller_id_name=Queue,origination_caller_id_number='fifo+RAFQ1'}{fifo_member_wait=nowait}user/1001 >>>> >>>> >>>> >>>> >>>> >>> caller_count="0" waiting_count="0" importance="0"> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> freeswitch at internal> 2010-06-24 13:42:14.673471 [NOTICE] >>>> switch_channel.c:776 New Channel sofia/external/22808182 at noProvider[4c530c08-7f96-11df-9204-6904a4602528] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4293 Channel >>>> sofia/external/22808182 at noProvider entering state [received][100] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4304 Remote SDP: >>>> v=0 >>>> o=- 6 2 IN IP4 192.168.128.31 >>>> s=CounterPath eyeBeam 1.5 >>>> c=IN IP4 192.168.128.31 >>>> t=0 0 >>>> m=audio 37558 RTP/AVP 107 0 8 18 101 >>>> a=rtpmap:107 BV32/16000 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-15 >>>> >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [BV32:107:16000:20]/[G7221:115:32000:20] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [BV32:107:16000:20]/[G7221:107:16000:20] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [BV32:107:16000:20]/[G722:9:8000:20] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [BV32:107:16000:20]/[PCMU:0:8000:20] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [BV32:107:16000:20]/[PCMA:8:8000:20] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [BV32:107:16000:20]/[GSM:3:8000:20] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [PCMU:0:8000:20]/[G7221:115:32000:20] >>>> 2010-06-24 13:42:14.673471 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/22808182 at noProvider) Running State Change CS_NEW >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [PCMU:0:8000:20]/[G7221:107:16000:20] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [PCMU:0:8000:20]/[G722:9:8000:20] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [PCMU:0:8000:20]/[PCMU:0:8000:20] >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:2462 Set Codec >>>> sofia/external/22808182 at noProvider PCMU/8000 20 ms 160 samples >>>> 2010-06-24 13:42:14.673471 [DEBUG] switch_core_state_machine.c:320 >>>> (sofia/external/22808182 at noProvider) State NEW >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf >>>> send/recv payload to 101 >>>> 2010-06-24 13:42:14.673471 [DEBUG] sofia.c:4451 >>>> (sofia/external/22808182 at noProvider) State Change CS_NEW -> CS_INIT >>>> 2010-06-24 13:42:14.673471 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/external/22808182 at noProvider [BREAK] >>>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/22808182 at noProvider) Running State Change CS_INIT >>>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:338 >>>> (sofia/external/22808182 at noProvider) State INIT >>>> 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:83 >>>> sofia/external/22808182 at noProvider SOFIA INIT >>>> 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:117 >>>> (sofia/external/22808182 at noProvider) State Change CS_INIT -> CS_ROUTING >>>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/external/22808182 at noProvider [BREAK] >>>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:338 >>>> (sofia/external/22808182 at noProvider) State INIT going to sleep >>>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/22808182 at noProvider) Running State Change CS_ROUTING >>>> 2010-06-24 13:42:14.674505 [DEBUG] switch_channel.c:1474 >>>> (sofia/external/22808182 at noProvider) Callstate Change DOWN -> RINGING >>>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:341 >>>> (sofia/external/22808182 at noProvider) State ROUTING >>>> 2010-06-24 13:42:14.674505 [DEBUG] switch_channel.c:1333 >>>> (sofia/external/22808182 at noProvider) Callstate Change RINGING -> ACTIVE >>>> 2010-06-24 13:42:14.674505 [DEBUG] mod_sofia.c:140 >>>> sofia/external/22808182 at noProvider SOFIA ROUTING >>>> 2010-06-24 13:42:14.674505 [DEBUG] switch_core_state_machine.c:77 >>>> sofia/external/22808182 at noProvider Standard ROUTING >>>> 2010-06-24 13:42:14.674505 [INFO] mod_dialplan_xml.c:331 Processing >>>> Afshin Afzali->1880 in context public >>>> Dialplan: sofia/external/22808182 at noProvider parsing [public->unloop] >>>> continue=false >>>> Dialplan: sofia/external/22808182 at noProvider Regex (PASS) [unloop] >>>> ${unroll_loops}(true) =~ /^true$/ break=on-false >>>> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [unloop] >>>> ${sip_looped_call}() =~ /^true$/ break=on-false >>>> Dialplan: sofia/external/22808182 at noProvider parsing >>>> [public->outside_call] continue=true >>>> Dialplan: sofia/external/22808182 at noProvider Absolute Condition >>>> [outside_call] >>>> Dialplan: sofia/external/22808182 at noProvider Action >>>> set(outside_call=true) >>>> Dialplan: sofia/external/22808182 at noProvider parsing >>>> [public->call_debug] continue=true >>>> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [call_debug] >>>> ${call_debug}(false) =~ /^true$/ break=never >>>> Dialplan: sofia/external/22808182 at noProvider parsing >>>> [public->public_extensions] continue=false >>>> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) >>>> [public_extensions] destination_number(1880) =~ /^(10[01][0-9])$/ >>>> break=on-false >>>> Dialplan: sofia/external/22808182 at noProvider parsing >>>> [public->public_did] continue=false >>>> Dialplan: sofia/external/22808182 at noProvider Regex (FAIL) [public_did] >>>> destination_number(1880) =~ /^(5551212)$/ break=on-false >>>> Dialplan: sofia/external/22808182 at noProvider parsing >>>> [public->pub1880_did] continue=false >>>> Dialplan: sofia/external/22808182 at noProvider Regex (PASS) [pub1880_did] >>>> destination_number(1880) =~ /^(1880)$/ break=on-false >>>> Dialplan: sofia/external/22808182 at noProvider Action >>>> set(domain_name=192.168.128.36) >>>> Dialplan: sofia/external/22808182 at noProvider Action >>>> set(sound_prefix=/usr/local/freeswitch/sounds/en/us/callie) >>>> Dialplan: sofia/external/22808182 at noProvider Action >>>> set(fifo_music=local_stream://moh) >>>> Dialplan: sofia/external/22808182 at noProvider Action >>>> set(origination_caller_id_name=AFSHIN) >>>> Dialplan: sofia/external/22808182 at noProvider Action >>>> set(origination_caller_id_number=22808182) >>>> Dialplan: sofia/external/22808182 at noProvider Action answer() >>>> Dialplan: sofia/external/22808182 at noProvider Action sleep(500) >>>> Dialplan: sofia/external/22808182 at noProvider Action >>>> playback(ivr/ivr-generic_greeting.wav) >>>> Dialplan: sofia/external/22808182 at noProvider Action fifo( >>>> RAFQ1 at 192.168.128.36 in) >>>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:119 >>>> (sofia/external/22808182 at noProvider) State Change CS_ROUTING -> >>>> CS_EXECUTE >>>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/external/22808182 at noProvider [BREAK] >>>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:341 >>>> (sofia/external/22808182 at noProvider) State ROUTING going to sleep >>>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/22808182 at noProvider) Running State Change CS_EXECUTE >>>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:348 >>>> (sofia/external/22808182 at noProvider) State EXECUTE >>>> 2010-06-24 13:42:14.675445 [DEBUG] mod_sofia.c:233 >>>> sofia/external/22808182 at noProvider SOFIA EXECUTE >>>> 2010-06-24 13:42:14.675445 [DEBUG] switch_core_state_machine.c:157 >>>> sofia/external/22808182 at noProvider Standard EXECUTE >>>> EXECUTE sofia/external/22808182 at noProvider set(outside_call=true) >>>> 2010-06-24 13:42:14.675445 [DEBUG] mod_dptools.c:843 >>>> sofia/external/22808182 at noProvider SET [outside_call]=[true] >>>> EXECUTE sofia/external/22808182 at noProviderset(domain_name=192.168.128.36) >>>> 2010-06-24 13:42:14.675445 [DEBUG] mod_dptools.c:843 >>>> sofia/external/22808182 at noProvider SET [domain_name]=[192.168.128.36] >>>> EXECUTE sofia/external/22808182 at noProviderset(sound_prefix=/usr/local/freeswitch/sounds/en/us/callie) >>>> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >>>> sofia/external/22808182 at noProvider SET >>>> [sound_prefix]=[/usr/local/freeswitch/sounds/en/us/callie] >>>> EXECUTE sofia/external/22808182 at noProviderset(fifo_music=local_stream://moh) >>>> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >>>> sofia/external/22808182 at noProvider SET >>>> [fifo_music]=[local_stream://moh] >>>> EXECUTE sofia/external/22808182 at noProviderset(origination_caller_id_name=AFSHIN) >>>> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >>>> sofia/external/22808182 at noProvider SET >>>> [origination_caller_id_name]=[AFSHIN] >>>> EXECUTE sofia/external/22808182 at noProviderset(origination_caller_id_number=22808182) >>>> 2010-06-24 13:42:14.676396 [DEBUG] mod_dptools.c:843 >>>> sofia/external/22808182 at noProvider SET >>>> [origination_caller_id_number]=[22808182] >>>> EXECUTE sofia/external/22808182 at noProvider answer() >>>> 2010-06-24 13:42:14.676396 [DEBUG] sofia_glue.c:2702 AUDIO RTP >>>> [sofia/external/22808182 at noProvider] 192.168.128.36 port 30436 -> >>>> 192.168.128.31 port 37558 codec: 0 ms: 20 >>>> 2010-06-24 13:42:14.676396 [DEBUG] switch_rtp.c:1408 Starting timer >>>> [soft] 160 bytes per 20ms >>>> 2010-06-24 13:42:14.678369 [DEBUG] sofia_glue.c:2912 Set 2833 dtmf send >>>> payload to 101 >>>> 2010-06-24 13:42:14.678369 [DEBUG] sofia_glue.c:2917 Set 2833 dtmf >>>> receive payload to 101 >>>> 2010-06-24 13:42:14.678369 [DEBUG] mod_sofia.c:667 Local SDP >>>> sofia/external/22808182 at noProvider: >>>> v=0 >>>> o=FreeSWITCH 1277356498 1277356499 IN IP4 192.168.128.36 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.128.36 >>>> t=0 0 >>>> m=audio 30436 RTP/AVP 0 101 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=silenceSupp:off - - - - >>>> a=ptime:20 >>>> a=sendrecv >>>> >>>> 2010-06-24 13:42:14.678369 [DEBUG] switch_core_session.c:647 Send signal >>>> sofia/external/22808182 at noProvider [BREAK] >>>> 2010-06-24 13:42:14.678369 [NOTICE] mod_dptools.c:746 Channel >>>> [sofia/external/22808182 at noProvider] has been answered >>>> EXECUTE sofia/external/22808182 at noProvider sleep(500) >>>> 2010-06-24 13:42:14.678369 [DEBUG] sofia.c:4293 Channel >>>> sofia/external/22808182 at noProvider entering state [completed][200] >>>> 2010-06-24 13:42:14.718466 [DEBUG] switch_rtp.c:2512 Correct ip/port >>>> confirmed. >>>> 2010-06-24 13:42:14.782161 [DEBUG] sofia.c:4293 Channel >>>> sofia/external/22808182 at noProvider entering state [ready][200] >>>> EXECUTE sofia/external/22808182 at noProviderplayback(ivr/ivr-generic_greeting.wav) >>>> 2010-06-24 13:42:15.178406 [DEBUG] switch_ivr_play_say.c:1161 Codec >>>> Activated L16 at 8000hz 1 channels 20ms >>>> 2010-06-24 13:42:21.018797 [DEBUG] switch_ivr_play_say.c:1468 done >>>> playing file >>>> EXECUTE sofia/external/22808182 at noProvider fifo(RAFQ1 at 192.168.128.36in) >>>> 2010-06-24 13:42:21.018797 [DEBUG] mod_local_stream.c:421 Opening Stream >>>> [moh/8000] 8000hz >>>> 2010-06-24 13:42:21.018797 [DEBUG] switch_ivr_play_say.c:1161 Codec >>>> Activated L16 at 8000hz 1 channels 20ms >>>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>>> string 0 = [execute_on_answer=unset fifo_hangup_check] >>>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>>> string 1 = [fifo_hangup_check=RAFQ1 at 192.168.128.36] >>>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>>> string 2 = [origination_caller_id_name=Queue] >>>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>>> string 3 = [origination_caller_id_number=fifo+RAFQ1] >>>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>>> string 4 = [fifo_member_wait=nowait] >>>> 2010-06-24 13:42:21.564836 [DEBUG] switch_ivr_originate.c:1954 variable >>>> string 0 = [presence_id=1001 at 192.168.128.36] >>>> 2010-06-24 13:42:21.564836 [NOTICE] switch_channel.c:776 New Channel >>>> sofia/internal/sip:1001 at 192.168.128.31:63820[506eaaa4-7f96-11df-9205-6904a4602528] >>>> 2010-06-24 13:42:21.566617 [DEBUG] mod_sofia.c:3883 (sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820) State Change CS_NEW -> CS_INIT >>>> 2010-06-24 13:42:21.566617 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>>> 2010-06-24 13:42:21.566617 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >>>> CS_INIT >>>> 2010-06-24 13:42:21.566617 [DEBUG] switch_core_state_machine.c:338 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State INIT >>>> 2010-06-24 13:42:21.566617 [DEBUG] mod_sofia.c:83 sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820 SOFIA INIT >>>> 2010-06-24 13:42:21.567514 [DEBUG] mod_sofia.c:117 (sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820) State Change CS_INIT -> CS_ROUTING >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:338 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State INIT going to >>>> sleep >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >>>> CS_ROUTING >>>> 2010-06-24 13:42:21.567514 [DEBUG] sofia.c:4293 Channel sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820 entering state [calling][0] >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_channel.c:1474 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) Callstate Change DOWN -> >>>> RINGING >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:341 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State ROUTING >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_channel.c:1333 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) Callstate Change RINGING >>>> -> ACTIVE >>>> 2010-06-24 13:42:21.567514 [DEBUG] mod_sofia.c:140 sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820 SOFIA ROUTING >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_ivr_originate.c:64 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State Change CS_ROUTING >>>> -> CS_CONSUME_MEDIA >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:341 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State ROUTING going to >>>> sleep >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >>>> CS_CONSUME_MEDIA >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:360 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State CONSUME_MEDIA >>>> 2010-06-24 13:42:21.567514 [DEBUG] switch_core_state_machine.c:360 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State CONSUME_MEDIA >>>> going to sleep >>>> 2010-06-24 13:42:21.673796 [INFO] sofia.c:662 sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820 Update Callee ID to "1001" <1001> >>>> 2010-06-24 13:42:21.675668 [DEBUG] sofia.c:4293 Channel sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820 entering state [proceeding][180] >>>> 2010-06-24 13:42:21.675668 [NOTICE] sofia.c:4365 Ring-Ready >>>> sofia/internal/sip:1001 at 192.168.128.31:63820! >>>> 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4293 Channel sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820 entering state [completing][200] >>>> 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4304 Remote SDP: >>>> v=0 >>>> o=- 6 2 IN IP4 192.168.128.31 >>>> s=CounterPath X-Lite 3.0 >>>> c=IN IP4 192.168.128.31 >>>> t=0 0 >>>> m=audio 63470 RTP/AVP 0 8 101 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-15 >>>> >>>> 2010-06-24 13:42:22.082684 [DEBUG] sofia.c:4293 Channel sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820 entering state [ready][200] >>>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [PCMU:0:8000:20]/[G7221:115:32000:20] >>>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [PCMU:0:8000:20]/[G7221:107:16000:20] >>>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [PCMU:0:8000:20]/[G722:9:8000:20] >>>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3877 Audio Codec Compare >>>> [PCMU:0:8000:20]/[PCMU:0:8000:20] >>>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:2462 Set Codec >>>> sofia/internal/sip:1001 at 192.168.128.31:63820 PCMU/8000 20 ms 160 >>>> samples >>>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:3810 Set 2833 dtmf send >>>> payload to 101 >>>> 2010-06-24 13:42:22.082684 [DEBUG] sofia_glue.c:2702 AUDIO RTP >>>> [sofia/internal/sip:1001 at 192.168.128.31:63820] 192.168.128.36 port >>>> 30704 -> 192.168.128.31 port 63470 codec: 0 ms: 20 >>>> 2010-06-24 13:42:22.083576 [DEBUG] switch_rtp.c:1408 Starting timer >>>> [soft] 160 bytes per 20ms >>>> 2010-06-24 13:42:22.084524 [DEBUG] sofia_glue.c:2912 Set 2833 dtmf send >>>> payload to 101 >>>> 2010-06-24 13:42:22.084524 [DEBUG] sofia_glue.c:2917 Set 2833 dtmf >>>> receive payload to 101 >>>> 2010-06-24 13:42:22.084524 [NOTICE] sofia.c:4851 Channel >>>> [sofia/internal/sip:1001 at 192.168.128.31:63820] has been answered >>>> 2010-06-24 13:42:22.084524 [DEBUG] switch_channel.c:2549 sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820 execute on answer: >>>> unset(fifo_hangup_check) >>>> EXECUTE sofia/internal/sip:1001 at 192.168.128.31:63820unset(fifo_hangup_check) >>>> 2010-06-24 13:42:22.084524 [DEBUG] mod_dptools.c:951 UNSET >>>> [fifo_hangup_check] >>>> 2010-06-24 13:42:22.084524 [DEBUG] switch_ivr_originate.c:3271 Originate >>>> Resulted in Success: [sofia/internal/sip:1001 at 192.168.128.31:63820] >>>> 2010-06-24 13:42:22.085421 [DEBUG] switch_ivr_originate.c:3271 Originate >>>> Resulted in Success: [sofia/internal/sip:1001 at 192.168.128.31:63820] >>>> 2010-06-24 13:42:22.085421 [DEBUG] mod_fifo.c:530 (sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820) State Change CS_CONSUME_MEDIA -> >>>> CS_EXECUTE >>>> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>>> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) Running State Change >>>> CS_EXECUTE >>>> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:348 >>>> (sofia/internal/sip:1001 at 192.168.128.31:63820) State EXECUTE >>>> 2010-06-24 13:42:22.085421 [DEBUG] mod_sofia.c:233 sofia/internal/ >>>> sip:1001 at 192.168.128.31:63820 SOFIA EXECUTE >>>> 2010-06-24 13:42:22.085421 [DEBUG] switch_core_state_machine.c:157 >>>> sofia/internal/sip:1001 at 192.168.128.31:63820 Standard EXECUTE >>>> EXECUTE sofia/internal/sip:1001 at 192.168.128.31:63820 fifo( >>>> RAFQ1 at 192.168.128.36 out nowait) >>>> 2010-06-24 13:42:22.138560 [DEBUG] switch_rtp.c:2512 Correct ip/port >>>> confirmed. >>>> 2010-06-24 13:42:22.619153 [DEBUG] switch_ivr_play_say.c:1468 done >>>> playing file >>>> 2010-06-24 13:42:22.619153 [DEBUG] mod_fifo.c:1097 >>>> (sofia/external/22808182 at noProvider) State Change CS_EXECUTE -> >>>> CS_HIBERNATE >>>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/external/22808182 at noProvider [BREAK] >>>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:348 >>>> (sofia/external/22808182 at noProvider) State EXECUTE going to sleep >>>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/22808182 at noProvider) Running State Change CS_HIBERNATE >>>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:363 >>>> (sofia/external/22808182 at noProvider) State HIBERNATE >>>> 2010-06-24 13:42:22.619153 [DEBUG] mod_sofia.c:214 >>>> sofia/external/22808182 at noProvider SOFIA HIBERNATE >>>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:220 >>>> sofia/external/22808182 at noProvider Standard HIBERNATE >>>> 2010-06-24 13:42:22.619153 [DEBUG] switch_core_state_machine.c:363 >>>> (sofia/external/22808182 at noProvider) State HIBERNATE going to sleep >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:1088 >>>> (sofia/external/22808182 at noProvider) State Change CS_HIBERNATE -> >>>> CS_CONSUME_MEDIA >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/external/22808182 at noProvider [BREAK] >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/22808182 at noProvider) Running State Change >>>> CS_CONSUME_MEDIA >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:360 >>>> (sofia/external/22808182 at noProvider) State CONSUME_MEDIA >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:647 Send signal >>>> sofia/external/22808182 at noProvider [BREAK] >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:665 >>>> sofia/external/22808182 at noProvider CUSTOM HOLD >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:647 Send signal >>>> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:360 >>>> (sofia/external/22808182 at noProvider) State CONSUME_MEDIA going to sleep >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_ivr_bridge.c:1183 >>>> (sofia/external/22808182 at noProvider) State Change CS_CONSUME_MEDIA -> >>>> CS_EXCHANGE_MEDIA >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/external/22808182 at noProvider [BREAK] >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/22808182 at noProvider) Running State Change >>>> CS_EXCHANGE_MEDIA >>>> 2010-06-24 13:42:22.638406 [DEBUG] switch_core_state_machine.c:351 >>>> (sofia/external/22808182 at noProvider) State EXCHANGE_MEDIA >>>> 2010-06-24 13:42:22.638406 [DEBUG] mod_sofia.c:538 SOFIA EXCHANGE_MEDIA >>>> 2010-06-24 13:42:22.639349 [DEBUG] switch_core_session.c:708 Send signal >>>> sofia/external/22808182 at noProvider [BREAK] >>>> 2010-06-24 13:42:22.639349 [DEBUG] switch_core_session.c:708 Send signal >>>> sofia/internal/sip:1001 at 192.168.128.31:63820 [BREAK] >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > > -- > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/27bf8365/attachment-0001.html From dswardstrom at remotelink.com Thu Jun 24 08:52:28 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Thu, 24 Jun 2010 10:52:28 -0500 (CDT) Subject: [Freeswitch-users] Erlang Examples Message-ID: <403828297.111.1277394748296.JavaMail.root@srvr12.remotelinkml.com> I have been using JavaScript to handle a Conferencing application that started with the conf-ivr.js example program but is significantly more complex. This has been fun even though I had never used JavaScript before this year. However, there are things that seem to not be possible using JavaScript. I need to interact with several web based applications for several reasons and also need to provide some time based interactions with FreeSwitch and/or artifacts (Database entries, files of recorded conferences, etc). We (RemoteLink) have decided that the best solution for this support is to use an Erlang program and mod_erlang_event. So now I need to learn another language. But one thing that I do not find one the FreeSWITCH site is any Erlang examples. Are there some sample programs available such as one that would look for a certain type of event and print it out? I have found some semi-samples in the freeswitch-users archives but am somewhat ambivalent about using any of these without permission. Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom From andrew at hijacked.us Thu Jun 24 09:08:28 2010 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 24 Jun 2010 12:08:28 -0400 Subject: [Freeswitch-users] Erlang Examples In-Reply-To: <403828297.111.1277394748296.JavaMail.root@srvr12.remotelinkml.com> References: <403828297.111.1277394748296.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <20100624160827.GC17555@hijacked.us> On Thu, Jun 24, 2010 at 10:52:28AM -0500, David Swardstrom wrote: > I have been using JavaScript to handle a Conferencing application that started > with the conf-ivr.js example program but is significantly more complex. > This has been fun even though I had never used JavaScript before this year. > > However, there are things that seem to not be possible using JavaScript. > I need to interact with several web based applications for several reasons > and also need to provide some time based interactions with FreeSwitch and/or > artifacts (Database entries, files of recorded conferences, etc). > > We (RemoteLink) have decided that the best solution for this support is > to use an Erlang program and mod_erlang_event. So now I need to learn > another language. > > But one thing that I do not find one the FreeSWITCH site is any Erlang examples. > Are there some sample programs available such as one that would look for > a certain type of event and print it out? > > I have found some semi-samples in the freeswitch-users archives but am somewhat > ambivalent about using any of these without permission. > The people at idapted posted this example: http://developer.idapted.com/2010/04/22/build-complex-freeswitch-ivr-in-erlang/ Also, you can look at the OpenACD code, as it uses mod_erlang_event extensively (that's what I wrote it for): http://github.com/Vagabond/OpenACD Look specifically at the freeeswitch files under src. Finally, there's a freeswitch.erl file included in the freeswitch tree that is a wrapper around the modules's API (OpenACD makes a lot of use of this module). Andrew From stephen at stephenjc.com Thu Jun 24 09:18:42 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Thu, 24 Jun 2010 12:18:42 -0400 Subject: [Freeswitch-users] sched_hangup no hangup cause In-Reply-To: References: Message-ID: as always its the simple things. Thank You. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print On Thu, Jun 24, 2010 at 10:36 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Tested this and it works fine on my test box: > > > > from the look of your example, you are spelling allotted as alotted (one l > instead of 2) which is probably your problem. > > > > > On Thu, Jun 24, 2010 at 8:10 AM, Stephen Cattaneo wrote: > >> i have "execute_on_answer=sched_hangup +" + calltimeoutsec + " >> alotted_timeout" on my bleg but after this hangups and i check >> blegsession.cause it contains NONE, if the bleg party just hangs up i do get >> normal_clearing. >> >> if there is something im doing wrong or if you need more info just let me >> know. >> >> Thanks, >> Stephen C >> -All of my email addresses go to the same place >> -Save Paper, think before you print >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/01eb0eda/attachment.html From jan.berger at video24.no Thu Jun 24 09:27:46 2010 From: jan.berger at video24.no (Jan Berger) Date: Thu, 24 Jun 2010 18:27:46 +0200 Subject: [Freeswitch-users] Erlang Examples In-Reply-To: <403828297.111.1277394748296.JavaMail.root@srvr12.remotelinkml.com> References: <403828297.111.1277394748296.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <02B90EAA972341FF95218F95AF4E57A5@dell9400> Hi, The OpenACD guys are writing the ACD in Erlang and integrating to FS, so you might find something there. --- I don't know the Ericsson Language that well myself, but having had a look at it I decided to stay away from this technology. What I am doing is writing IVR's in Java, C# or even C++ and I decided to use vxml/ccxml to bring IVR capability into the standard dev environment so I can deal with business in a proper language. Jan -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Swardstrom Sent: 24. juni 2010 17:52 To: freeswitch-users Subject: [Freeswitch-users] Erlang Examples I have been using JavaScript to handle a Conferencing application that started with the conf-ivr.js example program but is significantly more complex. This has been fun even though I had never used JavaScript before this year. However, there are things that seem to not be possible using JavaScript. I need to interact with several web based applications for several reasons and also need to provide some time based interactions with FreeSwitch and/or artifacts (Database entries, files of recorded conferences, etc). We (RemoteLink) have decided that the best solution for this support is to use an Erlang program and mod_erlang_event. So now I need to learn another language. But one thing that I do not find one the FreeSWITCH site is any Erlang examples. Are there some sample programs available such as one that would look for a certain type of event and print it out? I have found some semi-samples in the freeswitch-users archives but am somewhat ambivalent about using any of these without permission. Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 Thu Jun 24 09:58:03 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 24 Jun 2010 11:58:03 -0500 Subject: [Freeswitch-users] Erlang Examples In-Reply-To: <02B90EAA972341FF95218F95AF4E57A5@dell9400> References: <403828297.111.1277394748296.JavaMail.root@srvr12.remotelinkml.com> <02B90EAA972341FF95218F95AF4E57A5@dell9400> Message-ID: The guy who wrote mod_erlang_event and a developer of OpenACD is the same guy already here helping him namely Andrew. On Thu, Jun 24, 2010 at 11:27 AM, Jan Berger wrote: > Hi, The OpenACD guys are writing the ACD in Erlang and integrating to FS, > so > you might find something there. > > --- > > I don't know the Ericsson Language that well myself, but having had a look > at it I decided to stay away from this technology. > > What I am doing is writing IVR's in Java, C# or even C++ and I decided to > use vxml/ccxml to bring IVR capability into the standard dev environment so > I can deal with business in a proper language. > > Jan > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David > Swardstrom > Sent: 24. juni 2010 17:52 > To: freeswitch-users > Subject: [Freeswitch-users] Erlang Examples > > I have been using JavaScript to handle a Conferencing application that > started > with the conf-ivr.js example program but is significantly more complex. > This has been fun even though I had never used JavaScript before this year. > > However, there are things that seem to not be possible using JavaScript. > I need to interact with several web based applications for several reasons > and also need to provide some time based interactions with FreeSwitch > and/or > artifacts (Database entries, files of recorded conferences, etc). > > We (RemoteLink) have decided that the best solution for this support is > to use an Erlang program and mod_erlang_event. So now I need to learn > another language. > > But one thing that I do not find one the FreeSWITCH site is any Erlang > examples. > Are there some sample programs available such as one that would look for > a certain type of event and print it out? > > I have found some semi-samples in the freeswitch-users archives but am > somewhat > ambivalent about using any of these without permission. > > Regards, > David Swardstrom > (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > -- 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/20100624/a4abaf39/attachment.html From vkozak at abisoft.spb.ru Thu Jun 24 10:45:12 2010 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Thu, 24 Jun 2010 21:45:12 +0400 Subject: [Freeswitch-users] HOW SHOW CALLER_ID_NUMBER WITHOUT IP References: <2CA0C040-5D9E-495A-AFE8-A24963542DDD@freeswitch.org> Message-ID: FreeSwitch set in from_field its IP (FS_IP). I need specifid value in from_field. How can I set in callerid my value? ----- Original Message ----- From: "Brian West" To: Sent: Thursday, June 24, 2010 5:32 PM Subject: Re: [Freeswitch-users] HOW SHOW CALLER_ID_NUMBER WITHOUT IP > More than likely its your device thats doing this. Polycom does this also > if the IP in the to/from/rpid don't match the ip or hostname of the proxy > it'll display the IP in the callerid field. Its not us doing it. > > /b > > On Jun 24, 2010, at 6:47 AM, Kozak Vladimir wrote: > >> hi everybody. >> >> >> I have one problem. I need to show in phone caller_id_number without IP >> address or caller_id_number + domain without IP address. I use api >> originate command with origination_caller_id_name and >> origination_caller_id_number parameters. >> bgapi originate >> {origination_caller_id_name=125 at 123.12.13.14,origination_caller_id_number=123 at 123.12.13.14}[origination_uuid=6daa7b7e-97e4-4790-827d-44ff4f40fd18]sofia/internal/sip:1009 at 172.26.10.65:61802;rinstance=437cf350c3a4546f >> &park() >> >> FS cuts value of origination_caller_id_name parameter (delete specified >> ip) - it's ok. >> And FS deletes specified ip from value of origination_caller_id_number >> parameter and adds FS-IP - it's bad. >> >> how else can I show caller info? >> how can I show caller domain? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 brian at freeswitch.org Thu Jun 24 10:50:39 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 24 Jun 2010 12:50:39 -0500 Subject: [Freeswitch-users] HOW SHOW CALLER_ID_NUMBER WITHOUT IP In-Reply-To: References: <2CA0C040-5D9E-495A-AFE8-A24963542DDD@freeswitch.org> Message-ID: <0C3FF56C-7466-4863-AF9A-07620E52CB0A@freeswitch.org> Again the device you're working with is BUSTED if the IP is causing it to display it. You're better off fixing your device. If you MUST put what ever you want in the invite then you can set the variable sip_invite_domain. /b On Jun 24, 2010, at 12:45 PM, Kozak Vladimir wrote: > FreeSwitch set in from_field its IP (FS_IP). I need specifid value in > from_field. How can I set in callerid my value? > > > > ----- Original Message ----- > From: "Brian West" > To: > Sent: Thursday, June 24, 2010 5:32 PM > Subject: Re: [Freeswitch-users] HOW SHOW CALLER_ID_NUMBER WITHOUT IP > > >> More than likely its your device thats doing this. Polycom does this also >> if the IP in the to/from/rpid don't match the ip or hostname of the proxy >> it'll display the IP in the callerid field. Its not us doing it. >> >> /b > From larclap at yahoo.com Thu Jun 24 12:11:56 2010 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 24 Jun 2010 12:11:56 -0700 Subject: [Freeswitch-users] Invalid Application hash Message-ID: <00e601cb13d1$1d2b7130$57825390$@yahoo.com> Recently I've been getting an Invalid Application hash error. Any ideas on what might be causing this. I have not changed anything in the configuration for a long while. I am using git-c58efca 2010-06-24 11-52-49 -0500 on Centos 5.3. Pastebin is http://pastebin.freeswitch.org/13255. Thanks Lars From mrene_lists at avgs.ca Thu Jun 24 12:19:47 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 24 Jun 2010 15:19:47 -0400 Subject: [Freeswitch-users] Invalid Application hash In-Reply-To: <00e601cb13d1$1d2b7130$57825390$@yahoo.com> References: <00e601cb13d1$1d2b7130$57825390$@yahoo.com> Message-ID: <29873289-D970-4AB9-A563-2916E73FDAA9@avgs.ca> Load mod_hash (and mod_db if you need the db app) , mod_limit stuff was refactored away Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-06-24, at 3:11 PM, Lars Zeb wrote: > Recently I've been getting an Invalid Application hash error. Any ideas on > what might be causing this. I have not changed anything in the configuration > for a long while. > > I am using git-c58efca 2010-06-24 11-52-49 -0500 on Centos 5.3. > > Pastebin is http://pastebin.freeswitch.org/13255. > > Thanks Lars > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 macedoslm at gmail.com Thu Jun 24 15:10:58 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Thu, 24 Jun 2010 19:10:58 -0300 Subject: [Freeswitch-users] ESL Conference Message-ID: Hi, I'm using an Outbound ESL connection. In this scenario, how can I send a call into a conference room? Thanks, -- Samuel Macedo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/d3103d47/attachment.html From dujinfang at gmail.com Thu Jun 24 18:35:05 2010 From: dujinfang at gmail.com (Seven Du) Date: Fri, 25 Jun 2010 09:35:05 +0800 Subject: [Freeswitch-users] Erlang Examples In-Reply-To: References: <403828297.111.1277394748296.JavaMail.root@srvr12.remotelinkml.com> <02B90EAA972341FF95218F95AF4E57A5@dell9400> Message-ID: Welcome to the Erlang world. Erlang was initially designed to write telecom softwares and I found gen_fsm is easy to use and very clear to describe business logic. I'm the idapted person. I updated the post in the bottom to be more complete. Another vote to OpenACD because it is written by the author of mod_erlang_event. :) 2010/6/25 Anthony Minessale : > The guy who wrote mod_erlang_event and a developer of OpenACD is the same > guy already here helping him namely Andrew. > > > On Thu, Jun 24, 2010 at 11:27 AM, Jan Berger wrote: >> >> Hi, The OpenACD guys are writing the ACD in Erlang and integrating to FS, >> so >> you might find something there. >> >> --- >> >> I don't know the Ericsson Language that well myself, but having had a look >> at it I decided to stay away from this technology. >> >> What I am doing is writing IVR's in Java, C# or even C++ and I decided to >> use vxml/ccxml to bring IVR capability into the standard dev environment >> so >> I can deal with business in a proper language. >> >> Jan >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David >> Swardstrom >> Sent: 24. juni 2010 17:52 >> To: freeswitch-users >> Subject: [Freeswitch-users] Erlang Examples >> >> I have been using JavaScript to handle a Conferencing application that >> started >> with the conf-ivr.js example program but is significantly more complex. >> This has been fun even though I had never used JavaScript before this >> year. >> >> However, there are things that seem to not be possible using JavaScript. >> I need to interact with several web based applications for several reasons >> and also need to provide some time based interactions with FreeSwitch >> and/or >> artifacts (Database entries, files of recorded conferences, etc). >> >> We (RemoteLink) have decided that the best solution for this support is >> to use an Erlang program and mod_erlang_event. So now I need to learn >> another language. >> >> But one thing that I do not find one the FreeSWITCH site is any Erlang >> examples. >> Are there some sample programs available such as one that would look for >> a certain type of event and print it out? >> >> I have found some semi-samples in the freeswitch-users archives but am >> somewhat >> ambivalent about using any of these without permission. >> >> Regards, >> David Swardstrom >> (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > > > -- > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn From msc at freeswitch.org Thu Jun 24 18:48:44 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 24 Jun 2010 18:48:44 -0700 Subject: [Freeswitch-users] ESL Conference In-Reply-To: References: Message-ID: This worked for me... Then I used nc (netcat) per the wiki ( http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Using_Netcat): nc -v -l 127.0.0.1 8084 Then I called 10002 from a phone and got connected: Connection from 127.0.0.1 port 8084 [tcp/*] accepted Then: connect\n\n (system spits out lots of info) Then: sendmsg\n call-command: execute\n execute-app-name: transfer\n execute-app-arg: 3000\n\n Blammo! A call in conf: reeswitch at internal>conference list Conference 3000-10.15.0.91 (1 member rate: 8000) 2;sofia/internal/1001 at 10.15.0.91 ;3d6c16d8-73bc-4548-9df1-56cb6a63e6d0;1001;1001;hear|speak|floor;0;0;300 freeswitch at internal> You can transfer to an existing conference or a new one or whatever you want... -MC On Thu, Jun 24, 2010 at 3:10 PM, Samuel Macedo wrote: > Hi, > > I'm using an Outbound ESL connection. In this scenario, how can I send a > call into a conference room? > > Thanks, > -- > Samuel Macedo > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/d7e035cd/attachment.html From macedoslm at gmail.com Thu Jun 24 18:52:12 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Thu, 24 Jun 2010 22:52:12 -0300 Subject: [Freeswitch-users] ESL Conference In-Reply-To: References: Message-ID: I've found the problem. I was sending 'break' command before calling 'conference' async. Thanks, -- Samuel Macedo On 24 June 2010 19:10, Samuel Macedo wrote: > Hi, > > I'm using an Outbound ESL connection. In this scenario, how can I send a > call into a conference room? > > Thanks, > -- > Samuel Macedo > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/b9ab9cf7/attachment.html From steve at barrettsystems.com Thu Jun 24 18:54:40 2010 From: steve at barrettsystems.com (Steve Butterfield) Date: Thu, 24 Jun 2010 19:54:40 -0600 Subject: [Freeswitch-users] Internal Profile Not Valid Message-ID: I was wondering where to change the /default/default/? The really strange thing is that the system has corrected itself as of this evening without my intervention. Could I have a bad build? Is there a way to update freeswitch instead of a full re-install? I am using Ubuntu 10.04 Would it help to post logs? If so which ones. Thank you, Steve Steve Butterfield - steve at barrettsystems.com ---------- Forwarded message ---------- > From: Steven Ayre > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 24 Jun 2010 09:59:19 +0100 > Subject: Re: [Freeswitch-users] Internal Profile Not Valid > I think /default/default/ should just be /default/ > > On 24 June 2010 05:49, Steve Butterfield wrote: > >> I am new to using freeswitch but somehow, overnight my internal sip >> profile became invalid and freeswitch occupies 90+ % of my memory. I have >> looked over my install and everything seems to be in order. One thing that >> stands out during boot up is : >> >> 2010-06-23 21:48:15.225851 [DEBUG] sofia.c:1317 Creating agent for >> internal-ipv6 >> 2010-06-23 21:48:17.766457 [ERR] switch_xml.c:1297 Couldnt open >> /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such >> file or directory) >> 2010-06-23 21:48:17.817992 [ERR] switch_xml.c:1297 Couldnt open >> /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such >> file or directory) >> 2010-06-23 21:48:17.906682 [ERR] switch_xml.c:1297 Couldnt open >> /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such >> file or directory) >> 2010-06-23 21:48:17.967737 [ERR] switch_xml.c:1297 Couldnt open >> /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such >> file or directory) >> 2010-06-23 21:48:18.629030 [DEBUG] sofia.c:1353 Created agent for >> internal-ipv6 >> >> Any help would be appreciated. >> Steve >> >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/247a48a3/attachment-0001.html From dujinfang at gmail.com Thu Jun 24 19:41:37 2010 From: dujinfang at gmail.com (Seven Du) Date: Fri, 25 Jun 2010 10:41:37 +0800 Subject: [Freeswitch-users] event-lock in mod_erlang_event Message-ID: Hi, As I can see, You cannot do this in inbound erlang sendmsg(FS, uuid, playback, "1.wav"); sendmsg(FS, uuid, playback, "2.wav"); sendmsg(FS, uuid, transfer, "xxxxx because it's async, and it will play 2.wav immediately. 1) Sure if I know the length of 1.wav I can sendmsg(FS, uuid, playback, 1.wav sleep(3000 sendmsg(FS, uuid, playback, 2.wav 2) Or I could wait the execute_complete event which will be a little complicated According to http://wiki.freeswitch.org/wiki/Event_socket_outbound Is it possible to send a event-lock param to lock the message temporarily? like sendmsg(FS, UUID, App, Args, [{"event-lock", "true"}]). -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn From irmatov at gmail.com Thu Jun 24 21:32:44 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Fri, 25 Jun 2010 09:32:44 +0500 Subject: [Freeswitch-users] mod_erlang_event problem In-Reply-To: <20100624145749.GA17555@hijacked.us> References: <20100624145749.GA17555@hijacked.us> Message-ID: Hi, Andrew! On Thu, Jun 24, 2010 at 7:57 PM, Andrew Thompson wrote: >> My application is very simple: it just prints all events received from >> freeswitch. The problem is, that call is being terminated immidiately. >> As far as I can see, phonebooth:launch is called successfully, it >> returns a pid of a new process. This new process is still alive after >> the call is finished, and it does not receive any events from >> freeswitch (if it would, it would print them to screen). Freeswitch >> log tells me that erlang_outbound_function exits as soon as it gets >> new pid >> > > Yeah, I think this is a bug - Can you show me the code that's running? The code is very simple, just a gen_server that does nothing excepting printing info about events via io:format. See attached phonebooth.erl (I would paste it inline, but it would break formatting). > I think the bug is in the RPC mechanism vs the ! variant, unfortunately > I don't have a box with *both* freeswitch and erlang on it handy at the > moment - can you write a simple daemon that waits for the get_pid > messages and spawns on demand, as described on the wiki, and see if it > works that way (that's the way I usually do it). That's the way it is done on my other server. And it works there fine. But, also, on that server I also use RPC mechanism the same way I done it on new one, and it works there fine too. It's just older freeswitch and older erlang there. Don't know if that matters, or may be RPC works there simply because I also use registered process and get_pid. Ok, now I have switched to processing of get_pid messages: I have added spawner.erl (attached), which is being registered as phonebooth process, processes get_new messages and starts new process for each call. phonebooth.erl is not changed. Same problem: 2010-06-25 09:23:41.891156 [DEBUG] mod_erlang_event.c:1316 got pid! 2010-06-25 09:23:41.891156 [DEBUG] mod_erlang_event.c:1446 exit erlang_outbound_function Log is attached. I am ready to perform further testing/ debugging if you'll tell me what to try next. -- Timur Irmatov, xmpp:irmatov at jabber.ru -------------- next part -------------- A non-text attachment was scrubbed... Name: phonebooth.erl Type: text/x-erlang Size: 2374 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/65735743/attachment.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: spawner.erl Type: text/x-erlang Size: 777 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/65735743/attachment-0001.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: fs.log Type: text/x-log Size: 10475 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/65735743/attachment-0002.bin From sid at eltc.ru Fri Jun 25 01:41:42 2010 From: sid at eltc.ru (Sergey Scheglov) Date: Fri, 25 Jun 2010 15:41:42 +0700 Subject: [Freeswitch-users] Question about max-members Message-ID: <20100625154142.5d5bf5b3@shadow.elt> Hi All, max-members limits members in conference room or limits members using profile (i.e. default)? Regards Sergey Scheglov From mj at mjw.se Thu Jun 24 10:42:16 2010 From: mj at mjw.se (mattias jonsson) Date: Thu, 24 Jun 2010 19:42:16 +0200 Subject: [Freeswitch-users] Voice mail Message-ID: <000101cb13c4$96958450$1516e255@mj> Have freeswitch integrated voicemail? From ravi.kuru at callture.com Thu Jun 24 12:06:26 2010 From: ravi.kuru at callture.com (Ravi Kuru) Date: Thu, 24 Jun 2010 15:06:26 -0400 Subject: [Freeswitch-users] try to run mod_java but loading issue Message-ID: Hi, I try to run PhoneTest on freeswitch and I followed the instruction but I got this error on freeswitch.log file when i start the freeswitch. 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so 2010-06-24 15:01:47.096942 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** what was the issue? Ravi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100624/2fb349a7/attachment-0001.html From mayankshrivastava.15 at gmail.com Thu Jun 24 22:21:48 2010 From: mayankshrivastava.15 at gmail.com (Mayank Shrivastava) Date: Fri, 25 Jun 2010 10:51:48 +0530 Subject: [Freeswitch-users] Using mod_unimrcp Message-ID: Hi, I am an undergraduate student, presently working on an academic project as part of my intern. I need to run ASR using our Nuance Speech Server, on a number of .wav files that I have. I have tried using UniMRCP, but am running into problems, as i persistently get a Completion Cause Error number 006 on the files that I run the ASR on. I know that FreeSWITCH has the mod_unimrcp module built in, and I was wondering how I could use FreeSWITCH for the same. >From http://lists.freeswitch.org/pipermail/freeswitch-users/2010-June/059335.html I have understood how to build the mod_unimrcp mod, but how do I proceed? Thanks, Mayank From chaitanya at vivainfomedia.com Fri Jun 25 00:12:12 2010 From: chaitanya at vivainfomedia.com (Chaitanya Bhatt // Viva) Date: Fri, 25 Jun 2010 12:42:12 +0530 Subject: [Freeswitch-users] FreeSwitch Configuration : Help in SIP provider & gateway configuration Message-ID: Hey I have installed FreeSwitch successfully but not getting how to use it. I want to use FreeSwitch in Inbound/Outbound IVRS application. We will be using FreeSwitch with Sangoma Card & PRIs. In configuration section i am not getting SIP provider & gateway configuration. I am newbie in FreeSwitch, Can you please guide how to proceed with SIP provider & gateway configuration ? Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya at vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/82e811ec/attachment.html From harry at vangberg.name Fri Jun 25 03:07:14 2010 From: harry at vangberg.name (Harry Vangberg) Date: Fri, 25 Jun 2010 12:07:14 +0200 Subject: [Freeswitch-users] =?iso-8859-1?q?Looking_for_FreeSWITCH_user_in_?= =?iso-8859-1?q?=D6resund-area_=28Copenhagen/Malm=F6=29?= Message-ID: Hello I am looking for an experienced FreeSWITCH user/administrator in the greater ?resund area (Copenhagen/Malm?) to join an interesting project. Please let me know if you are interested or know anybody that are. Best regards, Harry Vangberg http://harry.vangberg.name From rupa at rupa.com Fri Jun 25 06:30:12 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 25 Jun 2010 08:30:12 -0500 Subject: [Freeswitch-users] Voice mail In-Reply-To: <000101cb13c4$96958450$1516e255@mj> References: <000101cb13c4$96958450$1516e255@mj> Message-ID: Yes On Thu, Jun 24, 2010 at 12:42 PM, mattias jonsson wrote: > Have freeswitch integrated voicemail? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/40b5fef0/attachment.html From daniel.neubert at solomo.de Fri Jun 25 06:31:57 2010 From: daniel.neubert at solomo.de (Daniel Neubert) Date: Fri, 25 Jun 2010 15:31:57 +0200 Subject: [Freeswitch-users] Voice mail In-Reply-To: <000101cb13c4$96958450$1516e255@mj> References: <000101cb13c4$96958450$1516e255@mj> Message-ID: <4C24AFCD.3080309@solomo.de> Have you tried reading the wiki? Best regards / Mit freundlichen Gr??en, Daniel Neubert On 24.06.2010 19:42, mattias jonsson wrote: > Have freeswitch integrated voicemail? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 david.ponzone at gmail.com Fri Jun 25 06:36:29 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Fri, 25 Jun 2010 15:36:29 +0200 Subject: [Freeswitch-users] FreeSwitch Configuration : Help in SIP provider & gateway configuration In-Reply-To: References: Message-ID: <5770C938-4D33-4487-9FC3-4AA059C33BE3@gmail.com> Chaitanya, as a start, I would really recommend reading the wiki (http://wiki.freeswitch.org ). The mailing-list is mainly targeted at people who spent some time reading the wiki and who played around with FS. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/06/2010 ? 09:12, Chaitanya Bhatt // Viva a ?crit : > Hey > > I have installed FreeSwitch successfully but not getting how to use > it. I want to use FreeSwitch in Inbound/Outbound IVRS application. > We will be using FreeSwitch with Sangoma Card & PRIs. In > configuration section i am not getting SIP provider & gateway > configuration. > I am newbie in FreeSwitch, Can you please guide how to proceed with > SIP provider & gateway configuration ? > > Incase of any further queries, Please feel free to mail me or > contact me on the numbers provided below. > > Thanks & Regards, > Chaitanya Bhatt > Software Engineer. > > Viva Infomedia Pvt. Ltd. > 242, Oshiwara Industrial Centre, > New Link Road, Opp. Oshiwara Bus Depot, > Goregaon West, Mumbai 400104. > > Direct: +91.22.40310356 > Board: +91.22.40310310 > Email : chaitanya at vivainfomedia.com > > Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging > India Awards 2009 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/bb842ae6/attachment.html From a.afzali2003 at gmail.com Fri Jun 25 06:39:57 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Fri, 25 Jun 2010 18:09:57 +0430 Subject: [Freeswitch-users] Using fifo_orbit_announce For On-hook Agent Message-ID: Hi FreeSWITCH, I want to use fifo_orbit_announce to play specific agent greeting to his caller (can not insert it to caller dialplan). As my agents are on-hook (they use extensions to login / logout of queues) , I don't know if I could set this variable in login extension! If I can not use this way, is there any other way? appreciate all comments, BEST, -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/4e43933c/attachment.html From rupa at rupa.com Fri Jun 25 06:47:55 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 25 Jun 2010 08:47:55 -0500 Subject: [Freeswitch-users] Voice mail In-Reply-To: <4C24AFCD.3080309@solomo.de> References: <000101cb13c4$96958450$1516e255@mj> <4C24AFCD.3080309@solomo.de> Message-ID: Daniel (and others), suggesting the wiki is a good idea, but try doing it without the attitude. Even better (and I could have done so) would be to provide a link into the wiki: http://wiki.freeswitch.org/wiki/Mod_voicemail Remember, some people coming to the mailing list are here for the first time. Let's keep that in mind and try to keep a welcome tone. On Fri, Jun 25, 2010 at 8:31 AM, Daniel Neubert wrote: > Have you tried reading the wiki? > > Best regards / Mit freundlichen Gr??en, > Daniel Neubert > > > > On 24.06.2010 19:42, mattias jonsson wrote: > > Have freeswitch integrated voicemail? > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/9f1265f4/attachment-0001.html From rupa at rupa.com Fri Jun 25 06:49:19 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 25 Jun 2010 08:49:19 -0500 Subject: [Freeswitch-users] FreeSwitch Configuration : Help in SIP provider & gateway configuration In-Reply-To: References: Message-ID: I'm a little confused. You say that you will be running without SIP (cards + PRI) but then ask about setting up SIP. Which configuration are you trying to do? On Fri, Jun 25, 2010 at 2:12 AM, Chaitanya Bhatt // Viva < chaitanya at vivainfomedia.com> wrote: > Hey > > I have installed FreeSwitch successfully but not getting how to use it. I > want to use FreeSwitch in Inbound/Outbound IVRS application. > We will be using FreeSwitch with Sangoma Card & PRIs. In configuration > section i am not getting SIP provider & gateway configuration. > I am newbie in FreeSwitch, Can you please guide how to proceed with SIP > provider & gateway configuration ? > > Incase of any further queries, Please feel free to mail me or contact me on > the numbers provided below. > > Thanks & Regards, > Chaitanya Bhatt > Software Engineer. > > Viva Infomedia Pvt. Ltd. > 242, Oshiwara Industrial Centre, > New Link Road, Opp. Oshiwara Bus Depot, > Goregaon West, Mumbai 400104. > > Direct: +91.22.40310356 > Board: +91.22.40310310 > Email : chaitanya at vivainfomedia.com > > Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India > Awards 2009 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/72fcb4a8/attachment.html From steveayre at gmail.com Fri Jun 25 07:13:41 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 25 Jun 2010 15:13:41 +0100 Subject: [Freeswitch-users] try to run mod_java but loading issue In-Reply-To: References: Message-ID: 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so Looks like it's a dependancy problem. Does that file exist, and are there any other errors just before that line? Are you also on 32bit or 64bit? Looks like it's trying to load a 32bit version (i386) which might be the problem if you're on 64bit FS. -Steve On 24 June 2010 20:06, Ravi Kuru wrote: > Hi, > > I try to run PhoneTest on freeswitch and I followed the instruction but I > got this error on freeswitch.log file when i start the freeswitch. > > 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading > /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so > 2010-06-24 15:01:47.096942 [CRIT] switch_loadable_module.c:882 Error > Loading module /usr/local/freeswitch/mod/mod_java.so > **Module load routine returned an error** > > what was the issue? > > Ravi > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/2d3ff5ab/attachment.html From andrew at hijacked.us Fri Jun 25 07:59:58 2010 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 25 Jun 2010 10:59:58 -0400 Subject: [Freeswitch-users] event-lock in mod_erlang_event In-Reply-To: References: Message-ID: <20100625145958.GD17555@hijacked.us> On Fri, Jun 25, 2010 at 10:41:37AM +0800, Seven Du wrote: > Hi, > > As I can see, You cannot do this in inbound erlang > > sendmsg(FS, uuid, playback, "1.wav"); > sendmsg(FS, uuid, playback, "2.wav"); > sendmsg(FS, uuid, transfer, "xxxxx > > because it's async, and it will play 2.wav immediately. > > > 1) Sure if I know the length of 1.wav I can > > sendmsg(FS, uuid, playback, 1.wav > sleep(3000 > sendmsg(FS, uuid, playback, 2.wav > > 2) Or I could wait the execute_complete event which will be a little complicated > > > According to http://wiki.freeswitch.org/wiki/Event_socket_outbound > > Is it possible to send a event-lock param to lock the message temporarily? like > > sendmsg(FS, UUID, App, Args, [{"event-lock", "true"}]). > Yes, event-lock works from erlang too (its implemented down in the core I think). I use it to prevent exactly this problem. Feel free to update the wiki to note this is available. Andrew From andrew at hijacked.us Fri Jun 25 08:06:22 2010 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 25 Jun 2010 11:06:22 -0400 Subject: [Freeswitch-users] mod_erlang_event problem In-Reply-To: References: <20100624145749.GA17555@hijacked.us> Message-ID: <20100625150622.GE17555@hijacked.us> On Fri, Jun 25, 2010 at 09:32:44AM +0500, Timur Irmatov wrote: > Hi, Andrew! > That's the way it is done on my other server. And it works there fine. > But, also, on that server I also use RPC mechanism the same way I done > it on new one, and it works there fine too. It's just older freeswitch > and older erlang there. Don't know if that matters, or may be RPC > works there simply because I also use registered process and get_pid. So you use both mechanisms on a box with older freeswitch and erlang and both work fine? Can you narrow it down by matching versions on one side (freeswitch might be best) and see if the bug goes away/comes back? Its possible the new erlang releases have changed something in ei, but its more likely that I broke something in FreeSWITCH. What are the versions of FS/erlang on the box that works? Andrew From msc at freeswitch.org Fri Jun 25 10:23:49 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jun 2010 10:23:49 -0700 Subject: [Freeswitch-users] Question about max-members In-Reply-To: <20100625154142.5d5bf5b3@shadow.elt> References: <20100625154142.5d5bf5b3@shadow.elt> Message-ID: For any conference created based in this profile it limits the total number of conference participants. So if your conference profile has max-members=5 then any conference you create based on that profile will allow only 5 people to join the conference. -MC On Fri, Jun 25, 2010 at 1:41 AM, Sergey Scheglov wrote: > > Hi All, > > max-members limits members in conference room or limits members using > profile (i.e. default)? > > Regards > Sergey Scheglov > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/8d95c94e/attachment.html From stephen at mymessage.us Fri Jun 25 15:35:31 2010 From: stephen at mymessage.us (Stephen Cattaneo) Date: Fri, 25 Jun 2010 18:35:31 -0400 Subject: [Freeswitch-users] ringback answer vs preanswer Message-ID: i am trying to understand why when i do session.preanswer ringback works when i bridge the call but when i do session.answer it does not. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/7846cb62/attachment.html From msc at freeswitch.org Fri Jun 25 17:45:47 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jun 2010 17:45:47 -0700 Subject: [Freeswitch-users] ringback answer vs preanswer In-Reply-To: References: Message-ID: Because ringback occurs in early media. You need transfer ringback after the call is answered. See this page for more info: http://wiki.freeswitch.org/wiki/Variable_transfer_ringback -MC On Fri, Jun 25, 2010 at 3:35 PM, Stephen Cattaneo wrote: > i am trying to understand why when i do session.preanswer ringback works > when i bridge the call but when i do session.answer it does not. > > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/0b5b717a/attachment.html From anthony.minessale at gmail.com Fri Jun 25 19:35:32 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 25 Jun 2010 21:35:32 -0500 Subject: [Freeswitch-users] Erlang Examples In-Reply-To: References: <403828297.111.1277394748296.JavaMail.root@srvr12.remotelinkml.com> <02B90EAA972341FF95218F95AF4E57A5@dell9400> Message-ID: Hey Seven, is JP still working with you guys? We haven't heard from him in a year or more... =D On Thu, Jun 24, 2010 at 8:35 PM, Seven Du wrote: > Welcome to the Erlang world. > > Erlang was initially designed to write telecom softwares and I found > gen_fsm is easy to use and very clear to describe business logic. I'm > the idapted person. I updated the post in the bottom to be more > complete. > > Another vote to OpenACD because it is written by the author of > mod_erlang_event. :) > > 2010/6/25 Anthony Minessale : > > The guy who wrote mod_erlang_event and a developer of OpenACD is the same > > guy already here helping him namely Andrew. > > > > > > On Thu, Jun 24, 2010 at 11:27 AM, Jan Berger > wrote: > >> > >> Hi, The OpenACD guys are writing the ACD in Erlang and integrating to > FS, > >> so > >> you might find something there. > >> > >> --- > >> > >> I don't know the Ericsson Language that well myself, but having had a > look > >> at it I decided to stay away from this technology. > >> > >> What I am doing is writing IVR's in Java, C# or even C++ and I decided > to > >> use vxml/ccxml to bring IVR capability into the standard dev environment > >> so > >> I can deal with business in a proper language. > >> > >> Jan > >> > >> -----Original Message----- > >> From: freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > David > >> Swardstrom > >> Sent: 24. juni 2010 17:52 > >> To: freeswitch-users > >> Subject: [Freeswitch-users] Erlang Examples > >> > >> I have been using JavaScript to handle a Conferencing application that > >> started > >> with the conf-ivr.js example program but is significantly more complex. > >> This has been fun even though I had never used JavaScript before this > >> year. > >> > >> However, there are things that seem to not be possible using JavaScript. > >> I need to interact with several web based applications for several > reasons > >> and also need to provide some time based interactions with FreeSwitch > >> and/or > >> artifacts (Database entries, files of recorded conferences, etc). > >> > >> We (RemoteLink) have decided that the best solution for this support is > >> to use an Erlang program and mod_erlang_event. So now I need to learn > >> another language. > >> > >> But one thing that I do not find one the FreeSWITCH site is any Erlang > >> examples. > >> Are there some sample programs available such as one that would look for > >> a certain type of event and print it out? > >> > >> I have found some semi-samples in the freeswitch-users archives but am > >> somewhat > >> ambivalent about using any of these without permission. > >> > >> Regards, > >> David Swardstrom > >> (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.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 > > > > > > > > -- > > 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 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100625/2016c7e4/attachment-0001.html From lakindia89 at gmail.com Fri Jun 25 21:03:40 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Sat, 26 Jun 2010 09:33:40 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: Dear Antony, Can you please give me some inputs regarding my douts? On Sat, Jun 19, 2010 at 11:38 AM, lakshmanan ganapathy wrote: > Can you please tell me what is meant by group confirm timeouts?, because I > thought group confirm timeouts means you are talking about > group_confirm_cancel_timeout only. I'm sorry if my understanding is wrong. > > > Here I've tested the group_confirm_cancel_timeout, and it didn't worked. > But the same solution has worked for > Phillip Jones > Refer > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg14207.html. > > > My scenario is, > When a caller calls, an ESL Outbound socket program will be executed. > The program will set group_confirm_key=exec and group_confirm_file=perl > script.pl > In the script.pl I'll get some digits from the user to get the > confirmation, if true I'll allow him to bridge with the caller. > > I've also used [leg_timeout=10]. What happens is, when the user answers the > call after 5 seconds, then while getting the password, the call gets hangup, > due to leg_timeout. > > To solve this, I've added group_confirm_cancel_timeout=true, but still the > other end got hangup immediately, once it reaches the leg_timeout. > > Here is my dialplan: > > > expression="^.*$"> > data="continue_on_fail=true"/> > data="bypass_media=false"/> > data="ignore_early_media=true"/> > data="exec_after_bridge_app=park"/> > > > > > > Here is the commands they I used in NC: > connect > > sendmsg > call-command: execute > execute-app-name: answer > > > sendmsg > call-command:execute > execute-app-name: set > execute-app-arg: group_confirm_key=exec > > sendmsg > call-command:execute > execute-app-name: set > execute-app-arg: group_confirm_file=perl /root/bridge.pl > > > sendmsg > call-command:execute > execute-app-name: set > execute-app-arg: group_confirm_cancel_timeout=true > > > sendmsg > call-command: execute > execute-app-name: bridge > execute-app-arg: > {group_confirm_cancel_timeout=true}[leg_timeout=10]user/1006 > > Here is the bridge.pl: > #!/usr/bin/perl > use freeswitch; > use Data::Dumper; > > our $session; > freeswitch::consoleLog("info","Goint to get the digits"); > # To simulate the scenario I used sleep here. > sleep(30); > 1; > > Here is the FreeSwitch Log. > http://pastebin.freeswitch.org/13220 > > Kindly provide me some inputs. > > On Fri, Jun 18, 2010 at 7:44 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> I said leg timeout beats the group confirm timeouts >> >> group_confirm_cancel_timeout is a whole different variable, when you set >> that to true it will stop all the timeouts as soon as you reach >> group_confirm execution >> >> {group_confirm_cancel_timeout=true}[leg_timeout=10]sofia/foo/foo at bar.com >> >> >> >> On Fri, Jun 18, 2010 at 12:50 AM, lakshmanan ganapathy < >> lakindia89 at gmail.com> wrote: >> >>> Dear Antony, >>> >>> Also in the leg_timeout wiki >>> http://wiki.freeswitch.org/wiki/Variable_leg_timeout, it is stated as >>> follows >>> >>> "If you are using group confirm then you can cancel the timeout by using >>> the group_confirm_cancel_timeoutchannel variable." >>> >>> >>> >>> On Thu, Jun 17, 2010 at 8:22 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> no there is no way, besides making both timeouts longer. >>>> you could file a feature request/bounty to ask for a feature to stop the >>>> leg timer when you reach the confirm. >>>> >>>> >>>> On Thu, Jun 17, 2010 at 4:23 AM, Nagalenoj H. wrote: >>>> >>>>> Anthony, >>>>> But, then there is no use. Am I right? Usually, we'll use the >>>>> group_confirm_cancel_timeout only when we need to override the leg_timeout. >>>>> But it happens in reverse in this case., >>>>> >>>>> I've tried using the group_confirm_cancel_timeout along with >>>>> call_timeout and things happening similar like setting leg_timout. >>>>> >>>>> Then, tried without setting leg_timeout and call_timeout explicitly. >>>>> * In this case if the callee doesn't picks the call, it >>>>> disconnects the leg in 30 secs. >>>>> * If he answers the call and the script continues to execute, >>>>> the leg is disconnected in 60 secs. >>>>> >>>>> What I need to do is, when the callee picks the call the leg_timeout >>>>> should not be accounted more and the leg shouldn't be disconnected because >>>>> of leg_timeout after that. >>>>> >>>>> Any other way of doing this?! >>>>> >>>>> >>>>> >>>>> On Tue, Jun 15, 2010 at 10:53 PM, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>>> leg timeout beats the group confirm timeouts >>>>>> >>>>>> >>>>>> On Tue, Jun 15, 2010 at 12:28 AM, Nagalenoj H. wrote: >>>>>> >>>>>>> Dear friends, >>>>>>> I've tried using the group_confirm_cancel_timeout channel >>>>>>> variable. I've written a testing script to get digits before bridging. But, >>>>>>> it doesn't seem to be working. >>>>>>> >>>>>>> My understanding after reading wiki is, >>>>>>> * When I dial [leg_timeout=10]user/1005, if he answers before >>>>>>> timeout and in the process of giving digits, then the call shouldn't be >>>>>>> disconnected after the leg_timeout secs (10 sec in the example). >>>>>>> >>>>>>> But, When I experiment it, the call is getting disconnected after 10 >>>>>>> seconds and it doesn't bother whether the callee has answered the >>>>>>> call(Started giving digits) or not answered at all. >>>>>>> >>>>>>> I've checked it with nc as follows, >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name: set >>>>>>> execute-app-arg: group_confirm_key=exec >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name: set >>>>>>> execute-app-arg: group_confirm_file=perl /root/confirm.pl >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name: set >>>>>>> execute-app-arg: group_confirm_cancel_timeout=1 >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name: bridge >>>>>>> execute-app-arg: [leg_timeout=10]user/1005 >>>>>>> >>>>>>> And here is the script, >>>>>>> >>>>>>> use freeswitch; >>>>>>> our $session; >>>>>>> my $digit; >>>>>>> >>>>>>> while(1) { >>>>>>> # Wait till response timeout for the first digit. >>>>>>> $digit = $session->getDigits(1, "", 10000); >>>>>>> freeswitch::consoleLog ("info","Digit>>".$digit."<<"); >>>>>>> >>>>>>> if (! $session->ready() ) { >>>>>>> freeswitch::consoleLog("info","Going to Exit\n"); >>>>>>> last; >>>>>>> } >>>>>>> if (defined $digit and $digit ne "" ) { >>>>>>> freeswitch::consoleLog("info","DTMF received: >>>>>>> $digit\n"); >>>>>>> if ($digit eq '#') { >>>>>>> return; >>>>>>> } >>>>>>> } >>>>>>> else { >>>>>>> freeswitch::consoleLog("info","Timeout\n"); >>>>>>> $session->hangup(); >>>>>>> } >>>>>>> } >>>>>>> 1; >>>>>>> >>>>>>> If my understanding is right then, I believe there is something wrong >>>>>>> with channel_variable. >>>>>>> >>>>>>> Kindly help me to resolve this. >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> Nagalenoj H. >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at 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 >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> Nagalenoj H. >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> >> >> -- >> 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100626/aa6f40e6/attachment-0001.html From irmatov at gmail.com Fri Jun 25 22:22:17 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Sat, 26 Jun 2010 10:22:17 +0500 Subject: [Freeswitch-users] mod_erlang_event problem In-Reply-To: <20100625150622.GE17555@hijacked.us> References: <20100624145749.GA17555@hijacked.us> <20100625150622.GE17555@hijacked.us> Message-ID: On Fri, Jun 25, 2010 at 8:06 PM, Andrew Thompson wrote: > On Fri, Jun 25, 2010 at 09:32:44AM +0500, Timur Irmatov wrote: >> Hi, Andrew! >> That's the way it is done on my other server. And it works there fine. >> But, also, on that server I also use RPC mechanism the same way I done >> it on new one, and it works there fine too. It's just older freeswitch >> and older erlang there. Don't know if that matters, or may be RPC >> works there simply because I also use registered process and get_pid. > > So you use both mechanisms on a box with older freeswitch and erlang and > both work fine? Can you narrow it down by matching versions on one side > (freeswitch might be best) and see if the bug goes away/comes back? Its > possible the new erlang releases have changed something in ei, but its > more likely that I broke something in FreeSWITCH. What are the versions > of FS/erlang on the box that works? Yes, both mechanisms work on other box. It is Debian Lenny, 64 bit. Erlang is installed from Debian packages, it is 12B03. As for freeswitch, it is a little bit difficult to tell its version. I was installing it from svn repository, and can't tell exactly what revision that was. May be 16041, but definitely before 17000. version command reports just 'FreeSWITCH Version 1.0.trunk (hacked)'. I'll try compiling 16041 on Red Hat machine and tell you the results. Thank you for your support. -- Timur Irmatov, xmpp:irmatov at jabber.ru From mkellem at vontoo.com Fri Jun 25 09:12:56 2010 From: mkellem at vontoo.com (Marc Kellem) Date: Fri, 25 Jun 2010 12:12:56 -0400 Subject: [Freeswitch-users] Javascript: inconsistent session.ready() result when originating a call Message-ID: I'm using Javascript to originate a new outbound call. The Session ready() method returns inconsistent results when the call is answered. When it returns false, the session's causecode value is 0. Am I doing something wrong here, or could it be a bug? A simple test script and log output is below. I'm executing the script from the console using jsrun. I've noticed that switch_core_state_machine.c logs a state change when ready() returns true. I'm not sure if this is significant, but it might indicate a race condition. ---- test script ---- new_session = new Session( "{ignore_early_media=true}sofia/internal/1000%192.168.1.10" ); if ( new_session.ready() ) { console_log( "info", "Call connected.\n" ); // play audio.... } else { // the causecode is 0 when this fails on a successful call attempt console_log("err", "Call not connected. Cause: " + new_session.cause + "[" + new_session.causecode + "]\n" ); } ---- call connected log ---- [NOTICE] sofia.c:4851 Channel [sofia/internal/1000] has been answered [DEBUG] switch_ivr_originate.c:3273 Originate Resulted in Success: [sofia/internal/1000] [DEBUG] mod_spidermonkey.c:2866 (sofia/internal/1000) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 [BREAK] [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000) Running State Change CS_SOFT_EXECUTE [INFO] originate.js:1 Call connected. [DEBUG] switch_core_state_machine.c:354 (sofia/internal/1000) State SOFT_EXECUTE [DEBUG] mod_sofia.c:544 SOFIA SOFT_EXECUTE [DEBUG] switch_core_state_machine.c:200 sofia/internal/1000 Standard SOFT_EXECUTE [DEBUG] switch_core_state_machine.c:354 (sofia/internal/1000) State SOFT_EXECUTE going to sleep ---- call not connected log ---- [NOTICE] sofia.c:4851 Channel [sofia/internal/1000] has been answered [DEBUG] switch_ivr_originate.c:3273 Originate Resulted in Success: [sofia/internal/1000] [DEBUG] mod_spidermonkey.c:2866 (sofia/internal/1000) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 [BREAK] [ERR] originate.js:1 Call not connected. Cause: NONE(0) [DEBUG] switch_channel.c:2261 (sofia/internal/1000) Callstate Change ACTIVE -> HANGUP [NOTICE] mod_spidermonkey.c:3066 Hangup sofia/internal/1000 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100625/1e90a0ee/attachment.html From raymondchan at commverge.com Sat Jun 26 00:47:45 2010 From: raymondchan at commverge.com (Raymond Chan) Date: Sat, 26 Jun 2010 15:47:45 +0800 (HKT) Subject: [Freeswitch-users] Use lua to replace a .wav file Message-ID: <20100626154745.BOA85868@mail.commverge.com> Hi, I am using freeswitch as call announcement purpose. Admin user can call in and record a sound clip and then updated the announcement sound clip. I created IVR menu and use the recording lua to record the sound clip (to record as temp...uuid...wav file). It is ok. However, I do not know how to replace the announcement sound clip after the user press a key in ivr to confirm update. I try to use lua to do it, but it seems that it do not have file replacement function. Am I correct? Thanks Raymond From stephen at mymessage.us Sat Jun 26 04:20:13 2010 From: stephen at mymessage.us (Stephen Cattaneo) Date: Sat, 26 Jun 2010 07:20:13 -0400 Subject: [Freeswitch-users] Javascript: inconsistent session.ready() result when originating a call In-Reply-To: References: Message-ID: I would do the following: try { new_session = new Session( "sofia/internal/1000%192.168.1.10" ); console_log( "info", "Call connected.\n" ); // play audio.... } catch (e) { // the causecode is 0 when this fails on a successful call attempt console_log("err", "Call not connected. Cause: " + new_session.cause + "[" + new_session.causecode + "]\n" ); } Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print On Fri, Jun 25, 2010 at 12:12 PM, Marc Kellem wrote: > I'm using Javascript to originate a new outbound call. The Session ready() > method returns inconsistent results when the call is answered. When it > returns false, the session's causecode value is 0. Am I doing something > wrong here, or could it be a bug? > > A simple test script and log output is below. I'm executing the script from > the console using jsrun. I've noticed that switch_core_state_machine.c logs > a state change when ready() returns true. I'm not sure if this is > significant, but it might indicate a race condition. > > > ---- test script ---- > > new_session = new Session( > "{ignore_early_media=true}sofia/internal/1000%192.168.1.10" ); > if ( new_session.ready() ) { > console_log( "info", "Call connected.\n" ); > // play audio.... > } > else { > // the causecode is 0 when this fails on a successful call attempt > console_log("err", "Call not connected. Cause: " + new_session.cause + "[" > + new_session.causecode + "]\n" ); > } > > > ---- call connected log ---- > > [NOTICE] sofia.c:4851 Channel [sofia/internal/1000] has been answered > [DEBUG] switch_ivr_originate.c:3273 Originate Resulted in Success: > [sofia/internal/1000] > [DEBUG] mod_spidermonkey.c:2866 (sofia/internal/1000) State Change > CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE > [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 [BREAK] > [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000) Running State > Change CS_SOFT_EXECUTE > [INFO] originate.js:1 Call connected. > [DEBUG] switch_core_state_machine.c:354 (sofia/internal/1000) State > SOFT_EXECUTE > [DEBUG] mod_sofia.c:544 SOFIA SOFT_EXECUTE > [DEBUG] switch_core_state_machine.c:200 sofia/internal/1000 Standard > SOFT_EXECUTE > [DEBUG] switch_core_state_machine.c:354 (sofia/internal/1000) State > SOFT_EXECUTE going to sleep > > > ---- call not connected log ---- > > [NOTICE] sofia.c:4851 Channel [sofia/internal/1000] has been answered > [DEBUG] switch_ivr_originate.c:3273 Originate Resulted in Success: > [sofia/internal/1000] > [DEBUG] mod_spidermonkey.c:2866 (sofia/internal/1000) State Change > CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE > [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 [BREAK] > [ERR] originate.js:1 Call not connected. Cause: NONE(0) > [DEBUG] switch_channel.c:2261 (sofia/internal/1000) Callstate Change ACTIVE > -> HANGUP > [NOTICE] mod_spidermonkey.c:3066 Hangup sofia/internal/1000 > [CS_SOFT_EXECUTE] [NORMAL_CLEARING] > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100626/46b2eae1/attachment.html From a.afzali2003 at gmail.com Sat Jun 26 10:09:21 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sat, 26 Jun 2010 21:39:21 +0430 Subject: [Freeswitch-users] How to Access To Bridged Seesion Message-ID: Hi FreeSWITCH, What is the preferred method for obtaining bridged session from the current one? I've just found the signal_bond variable. BEST, -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100626/f0c3baa8/attachment.html From fraserredmond at gmail.com Sat Jun 26 15:38:58 2010 From: fraserredmond at gmail.com (Fraser Redmond) Date: Sat, 26 Jun 2010 23:38:58 +0100 Subject: [Freeswitch-users] Build error - on linux, during make, near mod_spandsp Message-ID: Ubuntu Lucid (10.04) via nightly tarball. Configure seems to run fine, but it breaks partway through the make. This is the output right before it exits: Making all in src Creating mod_spandsp.la /usr/bin/ld: cannot find -ljpeg collect2: ld returned 1 exit status gcc -shared .libs/mod_spandsp_la-mod_spandsp.o .libs/mod_spandsp_la-udptl.o .libs/mod_spandsp_la-mod_spandsp_fax.o .libs/mod_spandsp_la-mod_spandsp_dsp.o .libs/mod_spandsp_la-mod_spandsp_codecs.o -Wl,--rpath -Wl,/usr/src/freeswitch-snapshot/.libs -Wl,--rpath -Wl,/usr/local/freeswitch/lib -L/usr/src/freeswitch-snapshot/libs/tiff-3.8.2/libtiff/.libs -lm -lz -ljpeg /usr/src/freeswitch-snapshot/.libs/libfreeswitch.so -L/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib -L/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib/.libs -L/usr/src/freeswitch-snapshot/libs/apr/.libs -L/usr/src/freeswitch-snapshot/libs/srtp /usr/src/freeswitch-snapshot/libs/spandsp/src/.libs/libspandsp.a -L/usr/src/freeswitch-snapshot/libs/tiff-3.8.2/libtiff /usr/src/freeswitch-snapshot/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -lssl -lcrypto -lncurses -Wl,-soname -Wl,mod_spandsp.so -o .libs/mod_spandsp.so make[4]: *** [mod_spandsp.la] Error 1 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 The output above that (and in the configure) looks ok, unless there's something subtle to look out for. I carried on with the make install the first time, just in case, but the mod's that come after spandsp (alphabetically) weren't created. Any ideas? Cheers, Fraser ps. I have the prerequisites listed on the first line of http://wiki.freeswitch.org/wiki/Installation_Guide#Debian_Linux installed. I've installed on a Jaunty Ubuntu on AWS several times before without problem (using same prerequisites and installation steps.) pps. I know the git version is preferred, but I generally don't need to be on the cutting edge - but prove me wrong in this instance if you want :-) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100626/a2bb9629/attachment-0001.html From william.suffill at gmail.com Sat Jun 26 17:16:12 2010 From: william.suffill at gmail.com (William Suffill) Date: Sat, 26 Jun 2010 20:16:12 -0400 Subject: [Freeswitch-users] Build error - on linux, during make, near mod_spandsp In-Reply-To: References: Message-ID: According to the error you posted you are missing libjpeg. I'd suggest sudo apt-get install libjpeg-dev and that should resolve it for you. Probably should edit the wiki to include libjpeg-dev in addition to libtiff4-dev Try that and update the list if you have any further issues. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100626/ec4faeda/attachment.html From steveu at coppice.org Sat Jun 26 19:46:42 2010 From: steveu at coppice.org (Steve Underwood) Date: Sun, 27 Jun 2010 10:46:42 +0800 Subject: [Freeswitch-users] Build error - on linux, during make, near mod_spandsp In-Reply-To: References: Message-ID: <4C26BB92.90807@coppice.org> On 06/27/2010 08:16 AM, William Suffill wrote: > According to the error you posted you are missing libjpeg. I'd suggest > sudo apt-get install libjpeg-dev and that should resolve it for you. > Probably should edit the wiki to include libjpeg-dev in addition to > libtiff4-dev > > Try that and update the list if you have any further issues. > > -- W > Shouldn't libtiff4-dev depend of libjpeg-dev? Surely some of the defines in the JPEG headers are needed for the libtiff stuff. Steve From dujinfang at gmail.com Sat Jun 26 21:57:14 2010 From: dujinfang at gmail.com (Seven Du) Date: Sun, 27 Jun 2010 12:57:14 +0800 Subject: [Freeswitch-users] Erlang Examples In-Reply-To: References: <403828297.111.1277394748296.JavaMail.root@srvr12.remotelinkml.com> <02B90EAA972341FF95218F95AF4E57A5@dell9400> Message-ID: Yes. But he was not following this list as close as me. And he spent more time on company management than before. He just spoke on RailsConf a few days ago and I think you will hear from him after I CC this mail to him. http://en.oreilly.com/rails2010/public/schedule/detail/14302 2010/6/26 Anthony Minessale : > Hey Seven, is JP still working with you guys? > We haven't heard from him in a year or more... =D > > On Thu, Jun 24, 2010 at 8:35 PM, Seven Du wrote: >> >> Welcome to the Erlang world. >> >> Erlang was initially designed to write telecom softwares and I found >> gen_fsm is easy to use and very clear to describe business logic. I'm >> the idapted person. I updated the post in the bottom to be more >> complete. >> >> Another vote to OpenACD ?because it is written by the author of >> mod_erlang_event. :) >> >> 2010/6/25 Anthony Minessale : >> > The guy who wrote mod_erlang_event and a developer of OpenACD is the >> > same >> > guy already here helping him namely Andrew. >> > >> > >> > On Thu, Jun 24, 2010 at 11:27 AM, Jan Berger >> > wrote: >> >> >> >> Hi, The OpenACD guys are writing the ACD in Erlang and integrating to >> >> FS, >> >> so >> >> you might find something there. >> >> >> >> --- >> >> >> >> I don't know the Ericsson Language that well myself, but having had a >> >> look >> >> at it I decided to stay away from this technology. >> >> >> >> What I am doing is writing IVR's in Java, C# or even C++ and I decided >> >> to >> >> use vxml/ccxml to bring IVR capability into the standard dev >> >> environment >> >> so >> >> I can deal with business in a proper language. >> >> >> >> Jan >> >> >> >> -----Original Message----- >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> >> David >> >> Swardstrom >> >> Sent: 24. juni 2010 17:52 >> >> To: freeswitch-users >> >> Subject: [Freeswitch-users] Erlang Examples >> >> >> >> I have been using JavaScript to handle a Conferencing application that >> >> started >> >> with the conf-ivr.js example program but is significantly more complex. >> >> This has been fun even though I had never used JavaScript before this >> >> year. >> >> >> >> However, there are things that seem to not be possible using >> >> JavaScript. >> >> I need to interact with several web based applications for several >> >> reasons >> >> and also need to provide some time based interactions with FreeSwitch >> >> and/or >> >> artifacts (Database entries, files of recorded conferences, etc). >> >> >> >> We (RemoteLink) have decided that the best solution for this support is >> >> to use an Erlang program and mod_erlang_event. So now I need to learn >> >> another language. >> >> >> >> But one thing that I do not find one the FreeSWITCH site is any Erlang >> >> examples. >> >> Are there some sample programs available such as one that would look >> >> for >> >> a certain type of event and print it out? >> >> >> >> I have found some semi-samples in the freeswitch-users archives but am >> >> somewhat >> >> ambivalent about using any of these without permission. >> >> >> >> Regards, >> >> David Swardstrom >> >> (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.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 >> > >> > >> > >> > -- >> > 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 >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Blog: http://www.dujinfang.com >> Proj: ?http://www.freeswitch.org.cn >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn From larclap at yahoo.com Sun Jun 27 10:38:50 2010 From: larclap at yahoo.com (Lars Zeb) Date: Sun, 27 Jun 2010 10:38:50 -0700 Subject: [Freeswitch-users] Invalid Application hash In-Reply-To: <29873289-D970-4AB9-A563-2916E73FDAA9@avgs.ca> References: <00e601cb13d1$1d2b7130$57825390$@yahoo.com> <29873289-D970-4AB9-A563-2916E73FDAA9@avgs.ca> Message-ID: <002401cb161f$9aaf0630$d00d1290$@yahoo.com> Mathieu, Thanks for the reply. I rebuilt, making sure that modules.conf.xml in the source directory was loading mod_hash (). But rebuilding (either 'make current' or '.configure; make; make install') did not build mod/mod_hash.so. So trying to to issue 'load mod_hash' in the console gives: 2010-06-27 07:42:59.265080 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_hash.so **/usr/local/freeswitch/mod/mod_hash.so: cannot open shared object file: No such file or directory** What am I missing? I used c58efca_2010-06-24 on Centos 5.3. Thanks, Lars -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Thursday, June 24, 2010 12:20 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid Application hash Load mod_hash (and mod_db if you need the db app) , mod_limit stuff was refactored away Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-06-24, at 3:11 PM, Lars Zeb wrote: > Recently I've been getting an Invalid Application hash error. Any > ideas on what might be causing this. I have not changed anything in > the configuration for a long while. > > I am using git-c58efca 2010-06-24 11-52-49 -0500 on Centos 5.3. > > Pastebin is http://pastebin.freeswitch.org/13255. > > Thanks Lars > > > > _______________________________________________ > FreeSWITCH-users 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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 rupa at rupa.com Sun Jun 27 12:05:01 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Sun, 27 Jun 2010 14:05:01 -0500 Subject: [Freeswitch-users] Invalid Application hash In-Reply-To: <002401cb161f$9aaf0630$d00d1290$@yahoo.com> References: <00e601cb13d1$1d2b7130$57825390$@yahoo.com> <29873289-D970-4AB9-A563-2916E73FDAA9@avgs.ca> <002401cb161f$9aaf0630$d00d1290$@yahoo.com> Message-ID: a) Please reread the message I posted here about what to do to migrate to the new limit/hash/db stuff. It has all the instructions. b) you need to modify your modules.conf not modules.conf.xml. modules.conf is used by the build system to know which modules to build and install. It is in your source root directory. add applications/hash applications/db make current restart fs This will leave you in "compatibility" mode for mod_limit. On Sun, Jun 27, 2010 at 12:38 PM, Lars Zeb wrote: > Mathieu, > > Thanks for the reply. > > I rebuilt, making sure that modules.conf.xml in the source directory was > loading mod_hash (). But rebuilding (either 'make > current' or '.configure; make; make install') did not build > mod/mod_hash.so. > > So trying to to issue 'load mod_hash' in the console gives: > 2010-06-27 07:42:59.265080 [CRIT] switch_loadable_module.c:926 Error > Loading > module /usr/local/freeswitch/mod/mod_hash.so > **/usr/local/freeswitch/mod/mod_hash.so: cannot open shared object file: No > such file or directory** > > What am I missing? I used c58efca_2010-06-24 on Centos 5.3. > > Thanks, Lars > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Mathieu > Rene > Sent: Thursday, June 24, 2010 12:20 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Invalid Application hash > > Load mod_hash (and mod_db if you need the db app) , mod_limit stuff was > refactored away > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 2010-06-24, at 3:11 PM, Lars Zeb wrote: > > > Recently I've been getting an Invalid Application hash error. Any > > ideas on what might be causing this. I have not changed anything in > > the configuration for a long while. > > > > I am using git-c58efca 2010-06-24 11-52-49 -0500 on Centos 5.3. > > > > Pastebin is http://pastebin.freeswitch.org/13255. > > > > Thanks Lars > > > > > > > > _______________________________________________ > > FreeSWITCH-users 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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100627/3b57a516/attachment.html From william.suffill at gmail.com Sun Jun 27 12:36:34 2010 From: william.suffill at gmail.com (William Suffill) Date: Sun, 27 Jun 2010 15:36:34 -0400 Subject: [Freeswitch-users] Build error - on linux, during make, near mod_spandsp In-Reply-To: <4C26BB92.90807@coppice.org> References: <4C26BB92.90807@coppice.org> Message-ID: http://packages.ubuntu.com/karmic/libtiff4-dev Appears u are correct. Not sure why the configure test souldn't find it however. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100627/a565a5c6/attachment.html From larclap at yahoo.com Sun Jun 27 17:17:44 2010 From: larclap at yahoo.com (Lars Zeb) Date: Sun, 27 Jun 2010 17:17:44 -0700 Subject: [Freeswitch-users] Invalid Application hash In-Reply-To: References: <00e601cb13d1$1d2b7130$57825390$@yahoo.com> <29873289-D970-4AB9-A563-2916E73FDAA9@avgs.ca> <002401cb161f$9aaf0630$d00d1290$@yahoo.com> Message-ID: <004001cb1657$545d5640$fd1802c0$@yahoo.com> Rupa, Thanks for clearing this up. I think you meant: applications/mod_hash applications/mod_db I did search the mailing list earlier but could not find anything on this topic. Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Sunday, June 27, 2010 12:05 PM To: freeswitch-users Subject: Re: [Freeswitch-users] Invalid Application hash a) Please reread the message I posted here about what to do to migrate to the new limit/hash/db stuff. It has all the instructions. b) you need to modify your modules.conf not modules.conf.xml. modules.conf is used by the build system to know which modules to build and install. It is in your source root directory. add applications/hash applications/db make current restart fs This will leave you in "compatibility" mode for mod_limit. On Sun, Jun 27, 2010 at 12:38 PM, Lars Zeb wrote: Mathieu, Thanks for the reply. I rebuilt, making sure that modules.conf.xml in the source directory was loading mod_hash (). But rebuilding (either 'make current' or '.configure; make; make install') did not build mod/mod_hash.so. So trying to to issue 'load mod_hash' in the console gives: 2010-06-27 07:42:59.265080 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_hash.so **/usr/local/freeswitch/mod/mod_hash.so: cannot open shared object file: No such file or directory** What am I missing? I used c58efca_2010-06-24 on Centos 5.3. Thanks, Lars -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Thursday, June 24, 2010 12:20 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Invalid Application hash Load mod_hash (and mod_db if you need the db app) , mod_limit stuff was refactored away Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-06-24, at 3:11 PM, Lars Zeb wrote: > Recently I've been getting an Invalid Application hash error. Any > ideas on what might be causing this. I have not changed anything in > the configuration for a long while. > > I am using git-c58efca 2010-06-24 11-52-49 -0500 on Centos 5.3. > > Pastebin is http://pastebin.freeswitch.org/13255. > > Thanks Lars > > > > _______________________________________________ > FreeSWITCH-users 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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100627/e68fd266/attachment-0001.html From mrene_lists at avgs.ca Sun Jun 27 18:27:44 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 27 Jun 2010 21:27:44 -0400 Subject: [Freeswitch-users] How to Access To Bridged Seesion In-Reply-To: References: Message-ID: <60274886-04A5-4B8F-B5DE-BC120A6058F1@avgs.ca> Hi, signal_bond will always work (thats whats being used internally). However, if you are getting events from the core, you may have more luck with Other-Leg-Unique-ID as the full list of channel variables isn't always included. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-06-26, at 1:09 PM, afshin afzali wrote: > Hi FreeSWITCH, > > What is the preferred method for obtaining bridged session from the current one? > I've just found the signal_bond variable. > > BEST, > -- afshin > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 sid at eltc.ru Sun Jun 27 19:17:50 2010 From: sid at eltc.ru (=?UTF-8?B?0KnQtdCz0LvQvtCyINCh0LXRgNCz0LXQuQ==?=) Date: Mon, 28 Jun 2010 09:17:50 +0700 Subject: [Freeswitch-users] Question about max-members In-Reply-To: References: <20100625154142.5d5bf5b3@shadow.elt> Message-ID: <20100628091750.27a02364@shadow.elt> ? Fri, 25 Jun 2010 10:23:49 -0700 Michael Collins wrote: > For any conference created based in this profile it limits the total > number of conference participants. So if your conference profile has > max-members=5 then any conference you create based on that profile > will allow only 5 people to join the conference. > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 for the detailed response, Michael Regards Sergey Scheglov From mcampbellsmith at gmail.com Sun Jun 27 22:37:38 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 28 Jun 2010 15:37:38 +1000 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> Message-ID: Hi All, I'm not really sure if I got a firm answer for this one. Is the only way to ensure that transcoding is performed is by using late-negotiation? Why aren't all my codecs sent in the INVITE message to the B-leg (extension 1020)? Thanks! On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith wrote: > FS version and codecs are shown below, but my config file are probably > quite old. ?But I guess they should still work? > > All codecs are loaded, and the call works if late negotiation is set > on profile internal. > As I wrote above: > The call setup is extension 1000 calls extension 1020 > 1. Extension 1000 calls with preferred codec PCMU. ?PCMU is chosen by > FS as the A-leg codec > 2. Extension 1020 only supports GSM codec. ?The call fails with Not > Acceptable Here. > > I forgot to write that Extension 1000 does not support GSM (I want to > force transcoding). ?Is that why FS is filtering out GSM on the b-leg? > > freeswitch at internal> version > FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500) > > freeswitch at internal> show codecs > type,name,ikey > codec,ADPCM (IMA),mod_voipcodecs > codec,G.711 alaw,CORE_PCM_MODULE > codec,G.711 ulaw,CORE_PCM_MODULE > codec,G.722,mod_voipcodecs > codec,G.723.1 6.3k,mod_g723_1 > codec,G.726 16k,mod_voipcodecs > codec,G.726 16k (AAL2),mod_voipcodecs > codec,G.726 24k,mod_voipcodecs > codec,G.726 24k (AAL2),mod_voipcodecs > codec,G.726 32k,mod_voipcodecs > codec,G.726 32k (AAL2),mod_voipcodecs > codec,G.726 40k,mod_voipcodecs > codec,G.726 40k (AAL2),mod_voipcodecs > codec,G.729,mod_com_g729 > codec,GSM,mod_voipcodecs > codec,H.261 Video (passthru),mod_h26x > codec,H.263 Video (passthru),mod_h26x > codec,H.263+ Video (passthru),mod_h26x > codec,H.263++ Video (passthru),mod_h26x > codec,H.264 Video (passthru),mod_h26x > codec,LPC-10,mod_voipcodecs > 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,Speex,mod_speex > > 25 total. > > > On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone wrote: >> Mark, >> I confirm that, as I wrote that wiki page (the early negotiation part) :) >> Can you really confirm your FS version ? >> The parameter you showed is old. >> codec-prefs has been replaced in SIP profiles by: >> ?? ? >> ?? ? >> David Ponzone ?Direction Technique >> email: david.ponzone at ipeva.fr >> tel: ? ? ?01 74 03 18 97 >> gsm: ? 06 66 98 76 34 >> Service Client?IPeva >> tel: ? ? ?0811 46 26 26 >> www.ipeva.fr? -? ?www.ipeva-studio.com >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce >> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. >> >> >> >> Le 23/06/2010 ? 14:43, Mark Campbell-Smith a ?crit : >> >> Check this good wiki page for how FS negotiates codecs (early >> negotiation default): >> http://wiki.freeswitch.org/wiki/Codec_Negotiation >> >> I have this set in my internal profile: >> ??? >> >> and as stated before vars.xml: >> >> >> >> Setting late negotiation works (thanks Sergey), but reading the wiki >> page, I see the following sentence, which I interpret that GSM should >> still be sent: >> When FS calls leg B, the list of codecs in outbound-codec-prefs for >> the SIP profile is reorganized by pushing the codec negotiated above >> for leg A at the top . If B does not accept any of the codecs, the >> calls fails, obviously. >> >> >> >> On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano >> wrote: >> >> I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? >> >> Wouldn't fs only get involved when its media server is referred to? >> >> If the "other endpoint" will only accept G729, doesn't that mean you >> >> need to change that endpoint to also accept G711 or also license G729 >> >> in FS? >> >> On 6/23/10, Mark Campbell-Smith wrote: >> >> Test Setup: >> >> vars.xml: >> >> ? >> >> ? >> >> The call setup is extension 1000 calls extension 1020 >> >> 1. Extension 1000 calls with preferred codec PCMU. ?PCMU is chosen by >> >> FS as the A-leg codec >> >> 2. Extension 1020 only supports GSM codec. ?The call fails with Not >> >> Acceptable Here. >> >> FS only offers G729 and PCMU to 1020. ?How do I change the number of >> >> codecs that are offered to an extension? ?I know I can change the >> >> order in the codec_prefs, but would prefer FS to offer all three >> >> codecs to an extension. >> >> ? ?m=audio 23662 RTP/AVP 0 18 101 13 >> >> ? ?a=rtpmap:0 PCMU/8000 >> >> ? ?a=rtpmap:18 G729/8000 >> >> ? ?a=rtpmap:101 telephone-event/8000 >> >> ? ?a=fmtp:101 0-16 >> >> ? ?a=rtpmap:13 CN/8000 >> >> ? ?a=ptime:30 >> >> Thanks >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> -- >> >> Sent from my mobile device >> >> ====================== >> >> Tony Graziano, Manager >> >> Telephone: 434.984.8430 >> >> sip: tgraziano at voice.myitdepartment.net >> >> Fax: 434.984.8431 >> >> Email: tgraziano at myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> >> Telephone: 434.984.8426 >> >> sip: helpdesk at voice.myitdepartment.net >> >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> >> http://www.myitdepartment.net/gethelp/ >> >> Why do mathematicians always confuse Halloween and Christmas? >> >> Because 31 Oct = 25 Dec. >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.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 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 david.ponzone at gmail.com Mon Jun 28 00:15:22 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 28 Jun 2010 09:15:22 +0200 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> Message-ID: Please, retry with a genuine config (the default one would be a wise choice). David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 28/06/2010 ? 07:37, Mark Campbell-Smith a ?crit : > Hi All, > > I'm not really sure if I got a firm answer for this one. Is the only > way to ensure that transcoding is performed is by using > late-negotiation? Why aren't all my codecs sent in the INVITE message > to the B-leg (extension 1020)? > > Thanks! > > On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith > wrote: >> FS version and codecs are shown below, but my config file are >> probably >> quite old. But I guess they should still work? >> >> All codecs are loaded, and the call works if late negotiation is set >> on profile internal. >> As I wrote above: >> The call setup is extension 1000 calls extension 1020 >> 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by >> FS as the A-leg codec >> 2. Extension 1020 only supports GSM codec. The call fails with Not >> Acceptable Here. >> >> I forgot to write that Extension 1000 does not support GSM (I want to >> force transcoding). Is that why FS is filtering out GSM on the b- >> leg? >> >> freeswitch at internal> version >> FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500) >> >> freeswitch at internal> show codecs >> type,name,ikey >> codec,ADPCM (IMA),mod_voipcodecs >> codec,G.711 alaw,CORE_PCM_MODULE >> codec,G.711 ulaw,CORE_PCM_MODULE >> codec,G.722,mod_voipcodecs >> codec,G.723.1 6.3k,mod_g723_1 >> codec,G.726 16k,mod_voipcodecs >> codec,G.726 16k (AAL2),mod_voipcodecs >> codec,G.726 24k,mod_voipcodecs >> codec,G.726 24k (AAL2),mod_voipcodecs >> codec,G.726 32k,mod_voipcodecs >> codec,G.726 32k (AAL2),mod_voipcodecs >> codec,G.726 40k,mod_voipcodecs >> codec,G.726 40k (AAL2),mod_voipcodecs >> codec,G.729,mod_com_g729 >> codec,GSM,mod_voipcodecs >> codec,H.261 Video (passthru),mod_h26x >> codec,H.263 Video (passthru),mod_h26x >> codec,H.263+ Video (passthru),mod_h26x >> codec,H.263++ Video (passthru),mod_h26x >> codec,H.264 Video (passthru),mod_h26x >> codec,LPC-10,mod_voipcodecs >> 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,Speex,mod_speex >> >> 25 total. >> >> >> On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone > > wrote: >>> Mark, >>> I confirm that, as I wrote that wiki page (the early negotiation >>> part) :) >>> Can you really confirm your FS version ? >>> The parameter you showed is old. >>> codec-prefs has been replaced in SIP profiles by: >>> >>> >>> David Ponzone Direction Technique >>> email: david.ponzone at ipeva.fr >>> tel: 01 74 03 18 97 >>> gsm: 06 66 98 76 34 >>> Service Client IPeva >>> tel: 0811 46 26 26 >>> www.ipeva.fr - www.ipeva-studio.com >>> Ce message et toutes les pi?ces jointes sont confidentiels et >>> ?tablis ? >>> l'intention exclusive de ses destinataires. Toute utilisation ou >>> diffusion >>> non autoris?e est interdite. Tout message ?lectronique est >>> susceptible >>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce >>> message s'il >>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire >>> de ce >>> message, merci de le d?truire imm?diatement et d'avertir >>> l'exp?diteur. >>> >>> >>> >>> Le 23/06/2010 ? 14:43, Mark Campbell-Smith a ?crit : >>> >>> Check this good wiki page for how FS negotiates codecs (early >>> negotiation default): >>> http://wiki.freeswitch.org/wiki/Codec_Negotiation >>> >>> I have this set in my internal profile: >>> >>> >>> and as stated before vars.xml: >>> >>> >>> >>> Setting late negotiation works (thanks Sergey), but reading the wiki >>> page, I see the following sentence, which I interpret that GSM >>> should >>> still be sent: >>> When FS calls leg B, the list of codecs in outbound-codec-prefs for >>> the SIP profile is reorganized by pushing the codec negotiated above >>> for leg A at the top . If B does not accept any of the codecs, the >>> calls fails, obviously. >>> >>> >>> >>> On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano >>> wrote: >>> >>> I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? >>> >>> Wouldn't fs only get involved when its media server is referred to? >>> >>> If the "other endpoint" will only accept G729, doesn't that mean you >>> >>> need to change that endpoint to also accept G711 or also license >>> G729 >>> >>> in FS? >>> >>> On 6/23/10, Mark Campbell-Smith wrote: >>> >>> Test Setup: >>> >>> vars.xml: >>> >>> >>> >>> >> data="outbound_codec_prefs=G729,PCMU,GSM"/> >>> >>> The call setup is extension 1000 calls extension 1020 >>> >>> 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen >>> by >>> >>> FS as the A-leg codec >>> >>> 2. Extension 1020 only supports GSM codec. The call fails with Not >>> >>> Acceptable Here. >>> >>> FS only offers G729 and PCMU to 1020. How do I change the number of >>> >>> codecs that are offered to an extension? I know I can change the >>> >>> order in the codec_prefs, but would prefer FS to offer all three >>> >>> codecs to an extension. >>> >>> m=audio 23662 RTP/AVP 0 18 101 13 >>> >>> a=rtpmap:0 PCMU/8000 >>> >>> a=rtpmap:18 G729/8000 >>> >>> a=rtpmap:101 telephone-event/8000 >>> >>> a=fmtp:101 0-16 >>> >>> a=rtpmap:13 CN/8000 >>> >>> a=ptime:30 >>> >>> Thanks >>> >>> _______________________________________________ >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> -- >>> >>> Sent from my mobile device >>> >>> ====================== >>> >>> Tony Graziano, Manager >>> >>> Telephone: 434.984.8430 >>> >>> sip: tgraziano at voice.myitdepartment.net >>> >>> Fax: 434.984.8431 >>> >>> Email: tgraziano at myitdepartment.net >>> >>> LAN/Telephony/Security and Control Systems Helpdesk: >>> >>> Telephone: 434.984.8426 >>> >>> sip: helpdesk at voice.myitdepartment.net >>> >>> Fax: 434.984.8427 >>> >>> Helpdesk Contract Customers: >>> >>> http://www.myitdepartment.net/gethelp/ >>> >>> Why do mathematicians always confuse Halloween and Christmas? >>> >>> Because 31 Oct = 25 Dec. >>> >>> _______________________________________________ >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.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 >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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/20100628/c749a731/attachment-0001.html From a.afzali2003 at gmail.com Mon Jun 28 01:47:20 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Mon, 28 Jun 2010 12:17:20 +0330 Subject: [Freeswitch-users] How to Access To Bridged Seesion In-Reply-To: <60274886-04A5-4B8F-B5DE-BC120A6058F1@avgs.ca> References: <60274886-04A5-4B8F-B5DE-BC120A6058F1@avgs.ca> Message-ID: Hi Mathieu Thanks to reply, I don't know why I'm not able to get this variable !!! Maybe the name has changed ? BEST -- afshin On Mon, Jun 28, 2010 at 4:57 AM, Mathieu Rene wrote: > Hi, > > signal_bond will always work (thats whats being used internally). However, > if you are getting events from the core, you may have more luck with > Other-Leg-Unique-ID as the full list of channel variables isn't always > included. > > Cheers, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 2010-06-26, at 1:09 PM, afshin afzali wrote: > > > Hi FreeSWITCH, > > > > What is the preferred method for obtaining bridged session from the > current one? > > I've just found the signal_bond variable. > > > > BEST, > > -- afshin > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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/20100628/f9875117/attachment.html From erik.dekkers at wvds.nl Mon Jun 28 05:46:33 2010 From: erik.dekkers at wvds.nl (Erik Dekkers) Date: Mon, 28 Jun 2010 14:46:33 +0200 Subject: [Freeswitch-users] SIP/2.0 500 No session set by user Message-ID: Hi guys, I've got a problem with a Aastra SIP Phone. When I make a call to freeswitch from this Phone, I get an return SIP message "SIP/2.0 500 No session set by user". This occurs when I call from the Aastra Phone to a Cisco Skinny Phone (mod_skinny). When I call from Xlite to that same phone, it just Works. Besides that, when I call from the Aastra to Xlite and vice versa no problem occurs (SIP to SIP). I've searched the internet on this and came on the Sofia-Sip website. There I found this message is generated by soa_static.c (line 1131). There it says: if (user == NULL) return soa_set_status(ss, 500, "No session set by user"); But what does it mean by "USER"? Can somebody point me in the right direction so I can find out why it's replying with No session set by User Below i've attached the sip trace Thanks in advance. Regards, Erik (wvds-nl) =========================================================================== sip:111 at 192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKaa627edaa1360db77.0caf3c848d78869e6 Max-Forwards: 70 From: "Erik Dekkers" ;tag=ed6e722b28 To: "111" Call-ID: 86be725683a1a511 CSeq: 19626 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erik Dekkers" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 57i/2.6.0.66 Content-Type: application/sdp Content-Length: 236 v=0 o=MxSIP 0 0 IN IP4 192.168.10.204 s=SIP Call c=IN IP4 192.168.10.204 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv =========================================================================== SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKaa627edaa1360db77.0caf3c848d78869e6 From: "Erik Dekkers" ;tag=ed6e722b28 To: "111" Call-ID: 86be725683a1a511 CSeq: 19626 INVITE User-Agent: FreeSWITCH Rocks! Content-Length: 0 =========================================================================== SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKaa627edaa1360db77.0caf3c848d78869e6 From: "Erik Dekkers" ;tag=ed6e722b28 To: "111" ;tag=KgpQKj9UD29mj Call-ID: 86be725683a1a511 CSeq: 19626 INVITE User-Agent: FreeSWITCH Rocks! 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 Proxy-Authenticate: Digest realm="192.168.10.10", nonce="852fcd72-b282-df11-9043-00e081630bad", algorithm=MD5, qop="auth" Content-Length: 0 =========================================================================== ACK sip:111 at 192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKaa627edaa1360db77.0caf3c848d78869e6 Max-Forwards: 70 From: "Erik Dekkers" ;tag=ed6e722b28 To: "111" ;tag=KgpQKj9UD29mj Call-ID: 86be725683a1a511 CSeq: 19626 ACK User-Agent: Aastra 57i/2.6.0.66 Content-Length: 0 =========================================================================== INVITE sip:111 at 192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK97ad060aed9a928de.58ac06998b369e607 Proxy-Authorization: Digest username="201",realm="192.168.10.10",nonce="852fcd72-b282-df11-9043-00e081630bad",uri="sip:111 at 192.168.10.10:5060",response="b5416cff70cb923195c78c9c441d2186",algorithm=MD5,qop=auth,cnonce="deaeb147",nc=00000001 Max-Forwards: 70 From: "Erik Dekkers" ;tag=ed6e722b28 To: "111" Call-ID: 86be725683a1a511 CSeq: 19627 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erik Dekkers" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 57i/2.6.0.66 Content-Type: application/sdp Content-Length: 236 v=0 o=MxSIP 0 0 IN IP4 192.168.10.204 s=SIP Call c=IN IP4 192.168.10.204 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv =========================================================================== SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK97ad060aed9a928de.58ac06998b369e607 From: "Erik Dekkers" ;tag=ed6e722b28 To: "111" Call-ID: 86be725683a1a511 CSeq: 19627 INVITE User-Agent: FreeSWITCH Rocks! Content-Length: 0 =========================================================================== UPDATE sip:201 at 192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.10;rport;branch=z9hG4bKpSjty6aygUreN Max-Forwards: 70 From: "111" ;tag=mSFgNDtZaB07D To: "Erik Dekkers" ;tag=ed6e722b28 Call-ID: 86be725683a1a511 CSeq: 132746103 UPDATE Contact: User-Agent: FreeSWITCH Rocks! Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 P-Asserted-Identity: "test" =========================================================================== SIP/2.0 500 No session set by user Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK97ad060aed9a928de.58ac06998b369e607 From: "Erik Dekkers" ;tag=ed6e722b28 To: "111" ;tag=mSFgNDtZaB07D Call-ID: 86be725683a1a511 CSeq: 19627 INVITE Contact: User-Agent: FreeSWITCH Rocks! 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 =========================================================================== SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.10.10;rport=5060;branch=z9hG4bKpSjty6aygUreN;received=192.168.10.10 From: "111" ;tag=mSFgNDtZaB07D To: "Erik Dekkers" ;tag=ed6e722b28 Call-ID: 86be725683a1a511 CSeq: 132746103 UPDATE Server: Aastra 57i/2.6.0.66 Supported: gruu, path Content-Length: 0 =========================================================================== ACK sip:111 at 192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK97ad060aed9a928de.58ac06998b369e607 Max-Forwards: 70 From: "Erik Dekkers" ;tag=ed6e722b28 To: "111" ;tag=mSFgNDtZaB07D Call-ID: 86be725683a1a511 CSeq: 19627 ACK User-Agent: Aastra 57i/2.6.0.66 Content-Length: 0 From jeroeng at thegreek.com Sat Jun 26 17:57:29 2010 From: jeroeng at thegreek.com (Jeroen C. van Gelderen) Date: Sat, 26 Jun 2010 19:57:29 -0500 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? Message-ID: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> Hi everybody, I have a feeling I must be missing the obvious... I've been trying to get echo canceling to work with Freeswitch/FreeTDM/DAHDI but I have been failing miserably for days. When I enable echo cancellation (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes silent. In absence of better suggestions I was hoping to replicate someone's "known working" configuration. Is anyone successfully running the following combination: - any x86 hardware - any recent flavor Linux - any recent version of DAHDI using FXO ports - FreeSwitch/FreeTDM trunk from git - OSLEC echo canceller (or MG2) - (BONUS:) Xorcom Astribank with FXO/FXS ports. I addition to much Googling I've tried most permutations of: - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix - x86 and x64 single and multicore CPUs. - Non-PAE kernels for 32-bit installs. - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) - MG2 and OSLEC echo cancellers - Freeswitch from Git trunk In each case everything configures fine to the point that Asterisk 1.6 will function with echo cancellation enabled. So we know that DAHDI layer works. But Freeswitch channels go silent when echo cancellation is enabled. Puzzled, -Slim -- Jeroen C. "Slim" van Gelderen From roger_salloum at shaw.ca Sat Jun 26 07:53:31 2010 From: roger_salloum at shaw.ca (Roger Salloum) Date: Sat, 26 Jun 2010 07:53:31 -0700 Subject: [Freeswitch-users] Dialplan handling on call fails Message-ID: <96D730D0-57A6-4751-9194-1285B4DCC8D2@shaw.ca> Hi, I'm trying to setup a dialplan such that if one particular route fails it will try another. However, I do not want it to try another route once it had recieved a 180/183 in response from a gateway. I have not been able to determine how to accomplish this. For Example: So when 10001234567 is dialled i will match all 3. I'd like to be able to try 1, if failed, try 2, if failed try 3. All calls go out via an outbound proxy. Using the above examples if the gateway does not respond in time, the proxy generates a 408 REQUEST TIMEOUT error message. It will then fail out and try the next route. However, when the gateway responds with a 180/183 but there is no answer after 2 minutes the proxy, will generate a 480 NO ANSWER (also tried a 408 REQUEST TIMEOUT ). When Freeswitch receives this message it fails, and then attempts the third failure route. How do i prevent the dialplan from continuing once it has received a 180/183 when no one answers the phone? Thanks, From uzairsh at yahoo.com Mon Jun 28 04:16:28 2010 From: uzairsh at yahoo.com (Syed Hussain) Date: Mon, 28 Jun 2010 04:16:28 -0700 (PDT) Subject: [Freeswitch-users] IP SIP Trunk errors/issues Message-ID: <728083.6641.qm@web30502.mail.mud.yahoo.com> Hi , I have subscribed to sonovoip a sip trunk provider who uses IP based authentication for SIP services , I am trying to make outbound international calls which are being rejected by the provider for some reason, I am not able to figure out if there is something wrong with the configuration. I am listing out my configuration. Any help would be greatly appreciated . I have also tried #wiki.freeswitch.org/wiki/Provider_Configuration:_SonoVoIP configuration from freeswitch wiki gives errors . The voip configuration works perfectly with asterisk 1.6 My configuration is follows Under /usr/local/freeswitch/conf/sip_profiles/external cat sonovoip.xml My dialplan , under /usr/local/freeswitch/conf/dialplan/public.xml I did change the vars.xml ports The debug #pastebin.com/4MFhSWm9 Please advise if any other information is required. Thanks S From brian at freeswitch.org Mon Jun 28 06:06:01 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Jun 2010 08:06:01 -0500 Subject: [Freeswitch-users] SIP/2.0 500 No session set by user In-Reply-To: References: Message-ID: <54ACC057-79E9-4AEC-BB79-BD4761673321@freeswitch.org> Have you updated? I think we fixed this already but I can't tell unless you update and try the latest code from GIT. /b On Jun 28, 2010, at 7:46 AM, Erik Dekkers wrote: > Hi guys, > > I've got a problem with a Aastra SIP Phone. > > When I make a call to freeswitch from this Phone, I get an return SIP message "SIP/2.0 500 No session set by user". This occurs when I call from the Aastra Phone to a Cisco Skinny Phone (mod_skinny). > > When I call from Xlite to that same phone, it just Works. Besides that, when I call from the Aastra to Xlite and vice versa no problem occurs (SIP to SIP). > > I've searched the internet on this and came on the Sofia-Sip website. There I found this message is generated by soa_static.c (line 1131). There it says: > > if (user == NULL) > return soa_set_status(ss, 500, "No session set by user"); > > But what does it mean by "USER"? Can somebody point me in the right direction so I can find out why it's replying with No session set by User > > Below i've attached the sip trace > > Thanks in advance. > > Regards, > > Erik (wvds-nl) From engineerzuhairraza at gmail.com Mon Jun 28 06:12:49 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Mon, 28 Jun 2010 18:12:49 +0500 Subject: [Freeswitch-users] IP SIP Trunk errors/issues In-Reply-To: <728083.6641.qm@web30502.mail.mud.yahoo.com> References: <728083.6641.qm@web30502.mail.mud.yahoo.com> Message-ID: hi try this one On Mon, Jun 28, 2010 at 4:16 PM, Syed Hussain wrote: > Hi , > > I have subscribed to sonovoip a sip trunk provider who uses IP based > authentication for SIP services , > > I am trying to make outbound international calls which are being rejected > by the provider for some reason, I am not able to figure out if there is > something wrong with the configuration. > > I am listing out my configuration. Any help would be greatly appreciated . > I have also tried # > wiki.freeswitch.org/wiki/Provider_Configuration:_SonoVoIP configuration > from freeswitch wiki gives errors . > > The voip configuration works perfectly with asterisk 1.6 > > My configuration is follows > > Under > /usr/local/freeswitch/conf/sip_profiles/external > > cat sonovoip.xml > > > > > > > > > > > > > My dialplan , under /usr/local/freeswitch/conf/dialplan/public.xml > > > > > > > > I did change the vars.xml ports > > > > > > > > > > > > > > > The debug > > #pastebin.com/4MFhSWm9 > > > Please advise if any other information is required. > > Thanks > > S > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/3ab8105f/attachment.html From david.ponzone at gmail.com Mon Jun 28 06:35:35 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 28 Jun 2010 15:35:35 +0200 Subject: [Freeswitch-users] Dialplan handling on call fails In-Reply-To: <96D730D0-57A6-4751-9194-1285B4DCC8D2@shaw.ca> References: <96D730D0-57A6-4751-9194-1285B4DCC8D2@shaw.ca> Message-ID: If you use continue_on_fail, I don't think you need to set break="never". David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 26/06/2010 ? 16:53, Roger Salloum a ?crit : > Hi, > > I'm trying to setup a dialplan such that if one particular route > fails it will try another. However, I do not want it to try another > route once it had recieved a 180/183 in response from a gateway. I > have not been able to determine how to accomplish this. > > For Example: > > break="never"> > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > break="never"> > data="hangup_after_bridge=RECOVER_ON_TIMER_EXPIRE"/> > > > > break="never"> > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > So when 10001234567 is dialled i will match all 3. I'd like to be > able to try 1, if failed, try 2, if failed try 3. All calls go out > via an outbound proxy. > > Using the above examples if the gateway does not respond in time, > the proxy generates a 408 REQUEST TIMEOUT error message. It will > then fail out and try the next route. However, when the gateway > responds with a 180/183 but there is no answer after 2 minutes the > proxy, will generate a 480 NO ANSWER (also tried a 408 REQUEST > TIMEOUT ). When Freeswitch receives this message it fails, and then > attempts the third failure route. How do i prevent the dialplan from > continuing once it has received a 180/183 when no one answers the > phone? > > Thanks, > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/a18da2fc/attachment.html From david.ponzone at gmail.com Mon Jun 28 06:39:31 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 28 Jun 2010 15:39:31 +0200 Subject: [Freeswitch-users] IP SIP Trunk errors/issues In-Reply-To: <728083.6641.qm@web30502.mail.mud.yahoo.com> References: <728083.6641.qm@web30502.mail.mud.yahoo.com> Message-ID: <5D80E761-6CB7-4F00-8345-1EEF45AA7A10@gmail.com> Perhaps a network trace would help.... David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 28/06/2010 ? 13:16, Syed Hussain a ?crit : > Hi , > > I have subscribed to sonovoip a sip trunk provider who uses IP based > authentication for SIP services , > > I am trying to make outbound international calls which are being > rejected by the provider for some reason, I am not able to figure > out if there is something wrong with the configuration. > > I am listing out my configuration. Any help would be greatly > appreciated . > I have also tried #wiki.freeswitch.org/wiki/ > Provider_Configuration:_SonoVoIP configuration from freeswitch wiki > gives errors . > > The voip configuration works perfectly with asterisk 1.6 > > My configuration is follows > > Under > /usr/local/freeswitch/conf/sip_profiles/external > > cat sonovoip.xml > > > > > > > > > > > > > My dialplan , under /usr/local/freeswitch/conf/dialplan/public.xml > > > > > > > > I did change the vars.xml ports > > > > > > > > > > > > > > > The debug > > #pastebin.com/4MFhSWm9 > > > Please advise if any other information is required. > > Thanks > > S > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/b1fe49e0/attachment-0001.html From erik.dekkers at wvds.nl Mon Jun 28 06:47:56 2010 From: erik.dekkers at wvds.nl (Erik Dekkers) Date: Mon, 28 Jun 2010 15:47:56 +0200 Subject: [Freeswitch-users] SIP/2.0 500 No session set by user In-Reply-To: <54ACC057-79E9-4AEC-BB79-BD4761673321@freeswitch.org> References: <54ACC057-79E9-4AEC-BB79-BD4761673321@freeswitch.org> Message-ID: Hi Brian, Im running 2010-06-28 13:46:46.605940 [CONSOLE] switch_core.c:1607 FreeSWITCH Version 1.0.head (git-dc048ed 2010-06-27 19-33-11 -0400) right now and the problem still occurs. Any ideas? Regards, Erik -----Oorspronkelijk bericht----- Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Brian West Verzonden: Monday, June 28, 2010 3:06 PM Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] SIP/2.0 500 No session set by user Have you updated? I think we fixed this already but I can't tell unless you update and try the latest code from GIT. /b On Jun 28, 2010, at 7:46 AM, Erik Dekkers wrote: > Hi guys, > > I've got a problem with a Aastra SIP Phone. > > When I make a call to freeswitch from this Phone, I get an return SIP message "SIP/2.0 500 No session set by user". This occurs when I call from the Aastra Phone to a Cisco Skinny Phone (mod_skinny). > > When I call from Xlite to that same phone, it just Works. Besides that, when I call from the Aastra to Xlite and vice versa no problem occurs (SIP to SIP). > > I've searched the internet on this and came on the Sofia-Sip website. There I found this message is generated by soa_static.c (line 1131). There it says: > > if (user == NULL) > return soa_set_status(ss, 500, "No session set by user"); > > But what does it mean by "USER"? Can somebody point me in the right direction so I can find out why it's replying with No session set by User > > Below i've attached the sip trace > > Thanks in advance. > > Regards, > > Erik (wvds-nl) _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 mkellem at vontoo.com Mon Jun 28 06:51:13 2010 From: mkellem at vontoo.com (Marc Kellem) Date: Mon, 28 Jun 2010 09:51:13 -0400 Subject: [Freeswitch-users] Javascript: inconsistent session.ready() result when originating a call In-Reply-To: References: Message-ID: Stephen - I wrapped everything in a try/catch block like you suggested. The script does not throw an exception. On Sat, Jun 26, 2010 at 7:20 AM, Stephen Cattaneo wrote: > I would do the following: > > try > { > new_session = new Session( "sofia/internal/1000%192.168.1.10" ); > > > console_log( "info", "Call connected.\n" ); > // play audio.... > } > catch (e) > > { > // the causecode is 0 when this fails on a successful call attempt > console_log("err", "Call not connected. Cause: " + new_session.cause + "[" > + new_session.causecode + "]\n" ); > > } > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/12fdc0b7/attachment.html From stephen at stephenjc.com Mon Jun 28 06:56:37 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Mon, 28 Jun 2010 09:56:37 -0400 Subject: [Freeswitch-users] Javascript: inconsistent session.ready() result when originating a call In-Reply-To: References: Message-ID: are the logs any different? Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print On Mon, Jun 28, 2010 at 9:51 AM, Marc Kellem wrote: > Stephen - I wrapped everything in a try/catch block like you suggested. The > script does not throw an exception. > > On Sat, Jun 26, 2010 at 7:20 AM, Stephen Cattaneo wrote: > >> I would do the following: >> >> try >> { >> new_session = new Session( "sofia/internal/1000%192.168.1.10" ); >> >> >> console_log( "info", "Call connected.\n" ); >> // play audio.... >> } >> catch (e) >> >> { >> // the causecode is 0 when this fails on a successful call attempt >> console_log("err", "Call not connected. Cause: " + new_session.cause + >> "[" + new_session.causecode + "]\n" ); >> >> } >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/f72f0fef/attachment.html From anthony.minessale at gmail.com Mon Jun 28 07:09:03 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Jun 2010 09:09:03 -0500 Subject: [Freeswitch-users] Javascript: inconsistent session.ready() result when originating a call In-Reply-To: References: Message-ID: try latest GIT and see if it is any better. On Mon, Jun 28, 2010 at 8:56 AM, stephen at stephenjc wrote: > are the logs any different? > > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print > > > > On Mon, Jun 28, 2010 at 9:51 AM, Marc Kellem wrote: > >> Stephen - I wrapped everything in a try/catch block like you suggested. >> The script does not throw an exception. >> >> On Sat, Jun 26, 2010 at 7:20 AM, Stephen Cattaneo wrote: >> >>> I would do the following: >>> >>> try >>> { >>> new_session = new Session( "sofia/internal/1000%192.168.1.10" ); >>> >>> >>> console_log( "info", "Call connected.\n" ); >>> // play audio.... >>> } >>> catch (e) >>> >>> { >>> // the causecode is 0 when this fails on a successful call attempt >>> console_log("err", "Call not connected. Cause: " + new_session.cause + >>> "[" + new_session.causecode + "]\n" ); >>> >>> } >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100628/efd9e0a6/attachment.html From deya787 at gmail.com Mon Jun 28 07:22:25 2010 From: deya787 at gmail.com (Deya M) Date: Mon, 28 Jun 2010 17:22:25 +0300 Subject: [Freeswitch-users] Beginner Question Message-ID: Hi, After doing some changes in my config, I am now unable to make any calls, eg. extension to extension! Looking at the log files, I can see that calling 5000 (Demo) has a context of public instead of default (or internal) I am not sure which xml file should I change, and change it to what ? Appreciate your help, D:- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/2d6ebebe/attachment.html From testeador01 at gmail.com Mon Jun 28 08:01:56 2010 From: testeador01 at gmail.com (Milena) Date: Mon, 28 Jun 2010 10:01:56 -0500 Subject: [Freeswitch-users] Beginner Question In-Reply-To: References: Message-ID: Hello and welcome to the wonderful world of FreeSWITCH ;) When you post an issue like this, it is hard to know what is going on; start by using pastebin: http://pastebin.freeswitch.org Paste there any relevant information about your directory/dialplan and the logs of what happens when you make a call. Then reply to this thread with the link(s) to the information you pasted there. Also if you're just looking for a quick fix, you could just try rebuilding the default directory/dialplan and start making changes again according to what you need. If you want to do this, just run "make samples" from the freeswitch source directory (usually /usr/src/freeswitch). Be aware that by doing so, all changes you have made to the default configs will be lost, so if they are important: back up first!!. If you need some more help, just post again. -Milena 2010/6/28 Deya M > Hi, > > After doing some changes in my config, I am now unable to make any calls, > eg. extension to extension! > > Looking at the log files, I can see that calling 5000 (Demo) has a context > of public instead of default (or internal) > I am not sure which xml file should I change, and change it to what ? > > Appreciate your help, > > D:- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/36b14abb/attachment-0001.html From moises.silva at gmail.com Mon Jun 28 08:03:41 2010 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 28 Jun 2010 11:03:41 -0400 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? In-Reply-To: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> References: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> Message-ID: This may be of help: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055505.html Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Sat, Jun 26, 2010 at 8:57 PM, Jeroen C. van Gelderen < jeroeng at thegreek.com> wrote: > Hi everybody, > > I have a feeling I must be missing the obvious... > > I've been trying to get echo canceling to work with > Freeswitch/FreeTDM/DAHDI > but I have been failing miserably for days. When I enable echo cancellation > (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes > silent. > > In absence of better suggestions I was hoping to replicate someone's "known > working" configuration. Is anyone successfully running the following > combination: > > - any x86 hardware > - any recent flavor Linux > - any recent version of DAHDI using FXO ports > - FreeSwitch/FreeTDM trunk from git > - OSLEC echo canceller (or MG2) > - (BONUS:) Xorcom Astribank with FXO/FXS ports. > > I addition to much Googling I've tried most permutations of: > - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix > - x86 and x64 single and multicore CPUs. > - Non-PAE kernels for 32-bit installs. > - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) > - MG2 and OSLEC echo cancellers > - Freeswitch from Git trunk > > In each case everything configures fine to the point that Asterisk 1.6 will > function with echo cancellation enabled. So we know that DAHDI layer works. > But Freeswitch channels go silent when echo cancellation is enabled. > > Puzzled, > -Slim > > -- > Jeroen C. "Slim" van Gelderen > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/78d9e218/attachment.html From testeador01 at gmail.com Mon Jun 28 08:05:33 2010 From: testeador01 at gmail.com (Milena) Date: Mon, 28 Jun 2010 10:05:33 -0500 Subject: [Freeswitch-users] Invalid Application hash In-Reply-To: <004001cb1657$545d5640$fd1802c0$@yahoo.com> References: <00e601cb13d1$1d2b7130$57825390$@yahoo.com> <29873289-D970-4AB9-A563-2916E73FDAA9@avgs.ca> <002401cb161f$9aaf0630$d00d1290$@yahoo.com> <004001cb1657$545d5640$fd1802c0$@yahoo.com> Message-ID: For future references: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-June/059396.html -Milena 2010/6/27 Lars Zeb > > I did search the mailing list earlier but could not find anything on this > topic. > > > > Lars > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/6b138c87/attachment.html From mrene_lists at avgs.ca Mon Jun 28 09:32:59 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 28 Jun 2010 12:32:59 -0400 Subject: [Freeswitch-users] How to Access To Bridged Seesion In-Reply-To: References: <60274886-04A5-4B8F-B5DE-BC120A6058F1@avgs.ca> Message-ID: How are you trying? Give me some context Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-06-28, at 4:47 AM, afshin afzali wrote: > Hi Mathieu > > Thanks to reply, > I don't know why I'm not able to get this variable !!! > Maybe the name has changed ? > > BEST > -- afshin > > > On Mon, Jun 28, 2010 at 4:57 AM, Mathieu Rene wrote: > Hi, > > signal_bond will always work (thats whats being used internally). However, if you are getting events from the core, you may have more luck with Other-Leg-Unique-ID as the full list of channel variables isn't always included. > > Cheers, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 2010-06-26, at 1:09 PM, afshin afzali wrote: > > > Hi FreeSWITCH, > > > > What is the preferred method for obtaining bridged session from the current one? > > I've just found the signal_bond variable. > > > > BEST, > > -- afshin > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/79f06c8e/attachment.html From mkellem at vontoo.com Mon Jun 28 09:33:03 2010 From: mkellem at vontoo.com (Marc Kellem) Date: Mon, 28 Jun 2010 12:33:03 -0400 Subject: [Freeswitch-users] Javascript: inconsistent session.ready() result when originating a call In-Reply-To: References: Message-ID: I did a fresh checkout/install of the latest GIT. Now switch_core_sqldb.c is logging errors when I run the script. When you filter out those errors, ready() still returns false and causecode=0. 2010-06-28 12:14:13.653989 [INFO] sofia.c:662 sofia/internal/1000 Update Callee ID to "1000" <1000> 2010-06-28 12:14:13.721843 [ERR] switch_core_sqldb.c:730 SQL ERR: [select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','192.168.1.10',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='1000' and (sub_to_host='192.168.1.10' or presence_hosts like '%192.168.1.10%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)] library routine called out of sequence 2010-06-28 12:14:13.721843 [ERR] switch_core_sqldb.c:730 SQL ERR: [select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_CONSUME_MEDIA','unknown','192.168.1.10',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='1000' and (sub_to_host='192.168.1.10' or presence_hosts like '%192.168.1.10%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)] library routine called out of sequence 2010-06-28 12:14:13.886956 [NOTICE] sofia.c:4369 Ring-Ready sofia/internal/1000! 2010-06-28 12:14:15.819755 [ERR] switch_core_sqldb.c:411 SQL ERR [library routine called out of sequence] update sip_dialogs set state='confirmed',presence_id='1000 at 192.168.1.10',presence_data='' where uuid='313369b2-82d0-11df-84f0-078cbd7c22e4'; 2010-06-28 12:14:15.825582 [NOTICE] sofia.c:4855 Channel [sofia/internal/1000] has been answered 2010-06-28 12:14:15.825582 [ERR] originate.js:1 Call not connected. Cause: NONE[0] ... [lots of switch_core_sqldb.c errors ] ... On Mon, Jun 28, 2010 at 10:09 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try latest GIT and see if it is any better. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/d1870b96/attachment.html From anthony.minessale at gmail.com Mon Jun 28 09:44:49 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Jun 2010 11:44:49 -0500 Subject: [Freeswitch-users] Javascript: inconsistent session.ready() result when originating a call In-Reply-To: References: Message-ID: update again small hiccup in the revisions On Mon, Jun 28, 2010 at 11:33 AM, Marc Kellem wrote: > I did a fresh checkout/install of the latest GIT. Now switch_core_sqldb.c > is logging errors when I run the script. When you filter out those errors, > ready() still returns false and causecode=0. > > 2010-06-28 12:14:13.653989 [INFO] sofia.c:662 sofia/internal/1000 Update > Callee ID to "1000" <1000> > 2010-06-28 12:14:13.721843 [ERR] switch_core_sqldb.c:730 SQL ERR: [select > sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','192.168.1.10',sip_presence.status,sip_presence.rpid > from sip_subscriptions left join sip_presence on > (sip_subscriptions.sub_to_user=sip_presence.sip_user and > sip_subscriptions.sub_to_host=sip_presence.sip_host and > sip_subscriptions.profile_name=sip_presence.profile_name) where > sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and > sub_to_user='1000' and (sub_to_host='192.168.1.10' or presence_hosts like > '%192.168.1.10%') and (sip_subscriptions.profile_name = 'internal' or > sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)] library > routine called out of sequence > 2010-06-28 12:14:13.721843 [ERR] switch_core_sqldb.c:730 SQL ERR: [select > sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_CONSUME_MEDIA','unknown','192.168.1.10',sip_presence.status,sip_presence.rpid > from sip_subscriptions left join sip_presence on > (sip_subscriptions.sub_to_user=sip_presence.sip_user and > sip_subscriptions.sub_to_host=sip_presence.sip_host and > sip_subscriptions.profile_name=sip_presence.profile_name) where > sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and > sub_to_user='1000' and (sub_to_host='192.168.1.10' or presence_hosts like > '%192.168.1.10%') and (sip_subscriptions.profile_name = 'internal' or > sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)] library > routine called out of sequence > 2010-06-28 12:14:13.886956 [NOTICE] sofia.c:4369 Ring-Ready > sofia/internal/1000! > 2010-06-28 12:14:15.819755 [ERR] switch_core_sqldb.c:411 SQL ERR [library > routine called out of sequence] > update sip_dialogs set state='confirmed',presence_id='1000 at 192.168.1.10',presence_data='' > where uuid='313369b2-82d0-11df-84f0-078cbd7c22e4'; > > 2010-06-28 12:14:15.825582 [NOTICE] sofia.c:4855 Channel > [sofia/internal/1000] has been answered > 2010-06-28 12:14:15.825582 [ERR] originate.js:1 Call not connected. Cause: > NONE[0] > ... [lots of switch_core_sqldb.c errors ] ... > > > > On Mon, Jun 28, 2010 at 10:09 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> try latest GIT and see if it is any better. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100628/207ef9f5/attachment-0001.html From a.afzali2003 at gmail.com Mon Jun 28 10:00:40 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Mon, 28 Jun 2010 21:30:40 +0430 Subject: [Freeswitch-users] How to Access To Bridged Seesion In-Reply-To: References: <60274886-04A5-4B8F-B5DE-BC120A6058F1@avgs.ca> Message-ID: I've test it on a call from public context which routes to user 1001 via a FIFO. I tried to get it from agent leg. Also by routing call from public context to 1001 which dials direct user number. BEST -- afshin On Mon, Jun 28, 2010 at 9:02 PM, Mathieu Rene wrote: > How are you trying? Give me some context > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 2010-06-28, at 4:47 AM, afshin afzali wrote: > > Hi Mathieu > > Thanks to reply, > I don't know why I'm not able to get this variable !!! > Maybe the name has changed ? > > BEST > -- afshin > > > On Mon, Jun 28, 2010 at 4:57 AM, Mathieu Rene wrote: > >> Hi, >> >> signal_bond will always work (thats whats being used internally). However, >> if you are getting events from the core, you may have more luck with >> Other-Leg-Unique-ID as the full list of channel variables isn't always >> included. >> >> Cheers, >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 2010-06-26, at 1:09 PM, afshin afzali wrote: >> >> > Hi FreeSWITCH, >> > >> > What is the preferred method for obtaining bridged session from the >> current one? >> > I've just found the signal_bond variable. >> > >> > BEST, >> > -- afshin >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.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 >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100628/b05d7717/attachment.html From mkellem at vontoo.com Mon Jun 28 10:08:42 2010 From: mkellem at vontoo.com (Marc Kellem) Date: Mon, 28 Jun 2010 13:08:42 -0400 Subject: [Freeswitch-users] Javascript: inconsistent session.ready() result when originating a call In-Reply-To: References: Message-ID: The sqldb errors are gone but the original problem still exists. Let me know if you need more information. Thanks for looking into it. -- Marc On Mon, Jun 28, 2010 at 12:44 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > update again small hiccup in the revisions > > > On Mon, Jun 28, 2010 at 11:33 AM, Marc Kellem wrote: > >> I did a fresh checkout/install of the latest GIT. Now switch_core_sqldb.c >> is logging errors when I run the script. When you filter out those errors, >> ready() still returns false and causecode=0. >> >> 2010-06-28 12:14:13.653989 [INFO] sofia.c:662 sofia/internal/1000 Update >> Callee ID to "1000" <1000> >> 2010-06-28 12:14:13.721843 [ERR] switch_core_sqldb.c:730 SQL ERR: [select >> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','192.168.1.10',sip_presence.status,sip_presence.rpid >> from sip_subscriptions left join sip_presence on >> (sip_subscriptions.sub_to_user=sip_presence.sip_user and >> sip_subscriptions.sub_to_host=sip_presence.sip_host and >> sip_subscriptions.profile_name=sip_presence.profile_name) where >> sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and >> sub_to_user='1000' and (sub_to_host='192.168.1.10' or presence_hosts like >> '%192.168.1.10%') and (sip_subscriptions.profile_name = 'internal' or >> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)] library >> routine called out of sequence >> 2010-06-28 12:14:13.721843 [ERR] switch_core_sqldb.c:730 SQL ERR: [select >> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_CONSUME_MEDIA','unknown','192.168.1.10',sip_presence.status,sip_presence.rpid >> from sip_subscriptions left join sip_presence on >> (sip_subscriptions.sub_to_user=sip_presence.sip_user and >> sip_subscriptions.sub_to_host=sip_presence.sip_host and >> sip_subscriptions.profile_name=sip_presence.profile_name) where >> sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and >> sub_to_user='1000' and (sub_to_host='192.168.1.10' or presence_hosts like >> '%192.168.1.10%') and (sip_subscriptions.profile_name = 'internal' or >> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)] library >> routine called out of sequence >> 2010-06-28 12:14:13.886956 [NOTICE] sofia.c:4369 Ring-Ready >> sofia/internal/1000! >> 2010-06-28 12:14:15.819755 [ERR] switch_core_sqldb.c:411 SQL ERR [library >> routine called out of sequence] >> update sip_dialogs set state='confirmed',presence_id='1000 at 192.168.1.10',presence_data='' >> where uuid='313369b2-82d0-11df-84f0-078cbd7c22e4'; >> >> 2010-06-28 12:14:15.825582 [NOTICE] sofia.c:4855 Channel >> [sofia/internal/1000] has been answered >> 2010-06-28 12:14:15.825582 [ERR] originate.js:1 Call not connected. Cause: >> NONE[0] >> ... [lots of switch_core_sqldb.c errors ] ... >> >> >> >> On Mon, Jun 28, 2010 at 10:09 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> try latest GIT and see if it is any better. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/1c1d8072/attachment.html From msc at freeswitch.org Mon Jun 28 10:24:14 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Jun 2010 10:24:14 -0700 Subject: [Freeswitch-users] Using fifo_orbit_announce For On-hook Agent In-Reply-To: References: Message-ID: I am not sure I understand what your setup is. Do you have a separate FIFO queue for each agent? That would be the only way that each agent could have his own specific announcement. Each FIFO queue has only one set of orbit parameters (extension, context, dialplan, announcement file). What are you trying to accomplish? -MC On Fri, Jun 25, 2010 at 6:39 AM, afshin afzali wrote: > Hi FreeSWITCH, > > I want to use fifo_orbit_announce to play specific agent greeting to his > caller (can not insert it to caller dialplan). As my agents are on-hook > (they use extensions to login / logout of queues) , I don't know if I could > set this variable in login extension! If I can not use this way, is there > any other way? > > appreciate all comments, > BEST, > -- afshin > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/f3d712bf/attachment-0001.html From msc at freeswitch.org Mon Jun 28 10:31:04 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Jun 2010 10:31:04 -0700 Subject: [Freeswitch-users] Use lua to replace a .wav file In-Reply-To: <20100626154745.BOA85868@mail.commverge.com> References: <20100626154745.BOA85868@mail.commverge.com> Message-ID: Do you have an IVR defined in XML? If so, did you issue "reloadxml" after the update? -MC On Sat, Jun 26, 2010 at 12:47 AM, Raymond Chan wrote: > Hi, > > I am using freeswitch as call announcement purpose. Admin user > can call in and > record a sound clip and then updated the announcement sound > clip. I created IVR > menu and use the recording lua to record the sound clip (to > record as > temp...uuid...wav file). It is ok. However, I do not know how > to replace the > announcement sound clip after the user press a key in ivr to > confirm update. I try > to use lua to do it, but it seems that it do not have file > replacement function. > Am I correct? > > Thanks > > Raymond > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/fdfa978b/attachment.html From mkellem at vontoo.com Mon Jun 28 10:37:33 2010 From: mkellem at vontoo.com (Marc Kellem) Date: Mon, 28 Jun 2010 13:37:33 -0400 Subject: [Freeswitch-users] Javascript: inconsistent session.ready() result when originating a call In-Reply-To: References: Message-ID: Anthony - I ported my test script to Lua and it has the same problem. newSession = freeswitch.Session("{ignore_early_media=true}sofia/ 192.168.1.10/1000") if newSession:ready() then freeswitch.consoleLog("info", "Call connected\n" ) else local cause = newSession:hangupCause() freeswitch.consoleLog("err", "Call not connected. Cause: " .. cause .. "\n" ) end -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/9c201c83/attachment.html From msc at freeswitch.org Mon Jun 28 10:40:35 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Jun 2010 10:40:35 -0700 Subject: [Freeswitch-users] IP SIP Trunk errors/issues In-Reply-To: <5D80E761-6CB7-4F00-8345-1EEF45AA7A10@gmail.com> References: <728083.6641.qm@web30502.mail.mud.yahoo.com> <5D80E761-6CB7-4F00-8345-1EEF45AA7A10@gmail.com> Message-ID: On Mon, Jun 28, 2010 at 6:39 AM, David Ponzone wrote: > Perhaps a network trace would help.... > > David is correct. Look at line #126 of your trace: 2010-06-27 18:08:38.787970 [DEBUG] switch_core_state_machine.c:53 sofia/external/6000 at X.X.X.X Standard REPORTING, cause: CALL_REJECTED You need to see the SIP traffic to get a better idea of why the carrier is rejecting your call. There are two primary ways of getting the trace: at the fs_cli using the command "sofia profile external siptrace on" or using tcpdump (or tshark, etc.) to get a pcap. More information available here: http://wiki.freeswitch.org/wiki/Packet_Capture -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/e2c5e55e/attachment.html From jonas.gauffin at gmail.com Mon Jun 28 10:51:33 2010 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Mon, 28 Jun 2010 19:51:33 +0200 Subject: [Freeswitch-users] Incompatible destination Message-ID: Hello, I get this SDP from one endpoint: v=0 o=- 40700591 0 IN IP4 130.244.X.XX s=Cisco SDP 0 c=IN IP4 130.244.X.XX t=0 0 m=audio 32512 RTP/AVP 8 0 99 102 101 a=rtpmap:99 G.729a/8000 a=rtpmap:102 G.729b/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 And this from the other end point: v=0 o=- 33485646 0 IN IP4 130.244.Y.YY s=Cisco SDP 0 c=IN IP4 130.244.Y.YY t=0 0 m=audio 17142 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 FreeSwitch gives me INCOMPATIBLE_DESTINATION. My trunk provider says that the SDP:s are valid. The first one looks OK by me. How about the second one? //Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/ae435884/attachment.html From david.ponzone at gmail.com Mon Jun 28 11:18:09 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 28 Jun 2010 20:18:09 +0200 Subject: [Freeswitch-users] Incompatible destination In-Reply-To: References: Message-ID: I don't remember if omission of rtpmap is acceptable. Can you check your FS configuration ? Perhap you don't accept 8 (PCMA) in the profile's inbound codecs ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 28/06/2010 ? 19:51, Jonas Gauffin a ?crit : > Hello, > > I get this SDP from one endpoint: > > v=0 > o=- 40700591 0 IN IP4 130.244.X.XX > s=Cisco SDP 0 > c=IN IP4 130.244.X.XX > t=0 0 > m=audio 32512 RTP/AVP 8 0 99 102 101 > a=rtpmap:99 G.729a/8000 > a=rtpmap:102 G.729b/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > And this from the other end point: > > v=0 > o=- 33485646 0 IN IP4 130.244.Y.YY > s=Cisco SDP 0 > c=IN IP4 130.244.Y.YY > t=0 0 > m=audio 17142 RTP/AVP 8 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > FreeSwitch gives me INCOMPATIBLE_DESTINATION. My trunk provider says > that the SDP:s are valid. The first one looks OK by me. How about > the second one? > > //Jonas > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/ba0235b3/attachment.html From msc at freeswitch.org Mon Jun 28 11:21:53 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Jun 2010 11:21:53 -0700 Subject: [Freeswitch-users] How to Access To Bridged Seesion In-Reply-To: References: <60274886-04A5-4B8F-B5DE-BC120A6058F1@avgs.ca> Message-ID: On Mon, Jun 28, 2010 at 10:00 AM, afshin afzali wrote: > I've test it on a call from public context which routes to user 1001 via a > FIFO. I tried to get it from agent leg. Also by routing call from public > context to 1001 which dials direct user number. > I think what Mathieu was asking for was to have some specific information. Can you pastebin your dialplan/scripts/etc. as well as a debug log? More information on how to collect useful information can be found here: http://wiki.freeswitch.org/wiki/Reporting_Bugs The first few sections of this page is all about collecting information and how to report it, even if you are not actually reporting a bug. Many "bugs" are really just configurations that need to be modified to produce the desired behavior. Having all of the information helps us in the debugging and troubleshooting processes. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/58f1d479/attachment-0001.html From brian at freeswitch.org Mon Jun 28 11:23:07 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Jun 2010 13:23:07 -0500 Subject: [Freeswitch-users] Incompatible destination In-Reply-To: References: Message-ID: Omission of the RTP map is fine if its a standard number like 0, 8, 18 and such that are assigned. What isn't valid is G.729a or G.729b both are 100% invalid. There is no such thing in the specs. /b On Jun 28, 2010, at 12:51 PM, Jonas Gauffin wrote: > FreeSwitch gives me INCOMPATIBLE_DESTINATION. My trunk provider says that the SDP:s are valid. The first one looks OK by me. How about the second one? > From david.ponzone at gmail.com Mon Jun 28 11:36:57 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 28 Jun 2010 20:36:57 +0200 Subject: [Freeswitch-users] Incompatible destination In-Reply-To: References: Message-ID: Note for later: never believe a provider when they say their packets are valid :) David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 28/06/2010 ? 19:51, Jonas Gauffin a ?crit : > Hello, > > I get this SDP from one endpoint: > > v=0 > o=- 40700591 0 IN IP4 130.244.X.XX > s=Cisco SDP 0 > c=IN IP4 130.244.X.XX > t=0 0 > m=audio 32512 RTP/AVP 8 0 99 102 101 > a=rtpmap:99 G.729a/8000 > a=rtpmap:102 G.729b/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > And this from the other end point: > > v=0 > o=- 33485646 0 IN IP4 130.244.Y.YY > s=Cisco SDP 0 > c=IN IP4 130.244.Y.YY > t=0 0 > m=audio 17142 RTP/AVP 8 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > FreeSwitch gives me INCOMPATIBLE_DESTINATION. My trunk provider says > that the SDP:s are valid. The first one looks OK by me. How about > the second one? > > //Jonas > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/e8f7d767/attachment.html From mark.maly at molcs.org Mon Jun 28 11:45:10 2010 From: mark.maly at molcs.org (Mark Maly) Date: Mon, 28 Jun 2010 13:45:10 -0500 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? In-Reply-To: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> References: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> Message-ID: <003e01cb16f2$09c7d920$1d578b60$@maly@molcs.org> I'm attempting to use Freeswitch/DAHDHI and OSLEC, as well. I agree in commenting out the "echocancellation" line in zt.conf. The channels seem to work OK but the echo cancellation does not seem to work (I can see the kernel drivers for it, but I still have a great deal of echo). Mark -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen C. van Gelderen Sent: Saturday, June 26, 2010 7:57 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? Hi everybody, I have a feeling I must be missing the obvious... I've been trying to get echo canceling to work with Freeswitch/FreeTDM/DAHDI but I have been failing miserably for days. When I enable echo cancellation (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes silent. In absence of better suggestions I was hoping to replicate someone's "known working" configuration. Is anyone successfully running the following combination: - any x86 hardware - any recent flavor Linux - any recent version of DAHDI using FXO ports - FreeSwitch/FreeTDM trunk from git - OSLEC echo canceller (or MG2) - (BONUS:) Xorcom Astribank with FXO/FXS ports. I addition to much Googling I've tried most permutations of: - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix - x86 and x64 single and multicore CPUs. - Non-PAE kernels for 32-bit installs. - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) - MG2 and OSLEC echo cancellers - Freeswitch from Git trunk In each case everything configures fine to the point that Asterisk 1.6 will function with echo cancellation enabled. So we know that DAHDI layer works. But Freeswitch channels go silent when echo cancellation is enabled. Puzzled, -Slim -- Jeroen C. "Slim" van Gelderen _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 fs-list at communicatefreely.net Mon Jun 28 11:45:12 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 28 Jun 2010 14:45:12 -0400 Subject: [Freeswitch-users] SIP header on only one fork of a bridge Message-ID: <4C28EDB8.6080609@communicatefreely.net> Hello list, I would like to bridge a call to multiple SIP endpoints, but add different headers to each. I'm not entirely sure how to do this. I have no problem exporting a SIP header that does what I want for one destination, but I'm not sure how to set it for two. My application is that I want two IP phones to ring - one with the internal ring-ring splash, the others with a group-answer (single ring, then lamp flash only), for administrative assistants, etc. How do I export different variables to each branch? Thanks! -Tim From djbinter at gmail.com Mon Jun 28 11:48:02 2010 From: djbinter at gmail.com (DJB International) Date: Mon, 28 Jun 2010 11:48:02 -0700 Subject: [Freeswitch-users] Question about Lua in diaplan Message-ID: I was trying to make lua to bridge in dialplan, but somehow it did not behave like XML dialplan: If I have this in XML: then, the second action will not be called if the first one hangs up the channel unless the first action failed. But, if I have this in Lua: session:execute("set","hangup_after_bridge=true") session:execute("set","continue_on_fail=true") for k, v in pairs(split(dst_gw_ip_list, ';')) do session:execute("bridge","sofia/gateway/" .. v .. "/" .. dest_out .. "") end then, the second action is also called. How do I make the lua script to behave like XML? Thank you, Dorn B. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/532f2abb/attachment.html From roger_salloum at shaw.ca Mon Jun 28 12:06:13 2010 From: roger_salloum at shaw.ca (Roger Salloum) Date: Mon, 28 Jun 2010 12:06:13 -0700 Subject: [Freeswitch-users] Dialplan handling on call fails In-Reply-To: References: <96D730D0-57A6-4751-9194-1285B4DCC8D2@shaw.ca> Message-ID: >From my understanding the break is used to handle what happens on the evaluation of the condition. I have break=never as I want it to try all rules for the extension even if the one above returns true or false. ----- Original Message ----- From: David Ponzone Date: Monday, June 28, 2010 6:36 am Subject: Re: [Freeswitch-users] Dialplan handling on call fails To: freeswitch-users at lists.freeswitch.org > If you use continue_on_fail, I don't think you need to set? > break="never". > > David Ponzone? Direction Technique > email: david.ponzone at ipeva.fr > tel:????? 01 74 03 18 97 > gsm:?? 06 66 98 76 34 > > Service Client IPeva > tel:????? 0811 46 26 26 > www.ipeva.fr? -?? www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis? > ? l'intention exclusive de ses destinataires. Toute utilisation > ou? > diffusion non autoris?e est interdite. Tout message ?lectronique > est? > susceptible d'alt?ration. IPeva d?cline toute responsabilit? > au? > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. > Si? > vous n'?tes pas destinataire de ce message, merci de le > d?truire? > imm?diatement et d'avertir l'exp?diteur. > > > > > Le 26/06/2010 ? 16:53, Roger Salloum a ?crit : > > > Hi, > > > > I'm trying to setup a dialplan such that if one particular > route? > > fails it will try another.? However, I do not want it to > try another? > > route once it had recieved a 180/183 in response from a > gateway. I? > > have not been able to determine how to accomplish this. > > > > For Example: > > > >?? expression="^(1000123.*)$"? > > break="never"> > >?????? application="set" data="hangup_after_bridge=true"/> > >? > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > >?????? application="bridge" data="sofia/gateway/carrier1/$1"/> > >?? > > > >?? expression="^(1000.*)$"? > > break="never"> > >?????? application="set"? > > data="hangup_after_bridge=RECOVER_ON_TIMER_EXPIRE"/> > >?????? application="bridge" data="sofia/gateway/carrier2/$1"/> > >?? > > > >?? expression="^(1.*)$"? > > break="never"> > >?????? application="set" data="hangup_after_bridge=true"/> > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > >???? data="sofia/gateway/carrier3/$1"/>>?? > > > > So when 10001234567 is dialled i will match all 3. I'd like to > be? > > able to try 1, if failed, try 2, if failed try 3. All calls go > out? > > via an outbound proxy. > > > > Using the above examples if the gateway does not respond in > time,? > > the proxy generates a 408 REQUEST TIMEOUT error message. It > will? > > then fail out and try the next route. However, when the > gateway? > > responds with a 180/183 but there is no answer after 2 minutes > the? > > proxy, will generate a 480 NO ANSWER (also tried a 408 > REQUEST? > > TIMEOUT ). When Freeswitch receives this message it fails, and > then? > > attempts the third failure route. How do i prevent the > dialplan from? > > continuing once it has received a 180/183 when no one answers > the? > > phone? > > > > Thanks, > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/df924365/attachment-0001.html From infos at madovsky.org Mon Jun 28 12:24:24 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 28 Jun 2010 15:24:24 -0400 Subject: [Freeswitch-users] How to Access To Bridged Seesion References: <60274886-04A5-4B8F-B5DE-BC120A6058F1@avgs.ca> Message-ID: Hi Mathieu, I tried to use Other-Leg-Unique-ID rather than signal_bond with this CLI command fs_cli -x "expand uuid_send_dtmf ${uuid_getvar[callerID] Other-Leg-Unique-ID)} 9 at 250" but doesn't work. signal_bond works well F ----- Original Message ----- From: "Mathieu Rene" To: Sent: Sunday, June 27, 2010 9:27 PM Subject: Re: [Freeswitch-users] How to Accesrs To Bridged Seesion > Hi, > > signal_bond will always work (thats whats being used internally). However, if you are getting events from the core, you may have more luck with Other-Leg-Unique-ID as the full list of channel variables isn't always included. > > Cheers, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 2010-06-26, at 1:09 PM, afshin afzali wrote: > >> Hi FreeSWITCH, >> >> What is the preferred method for obtaining bridged session from the current one? >> I've just found the signal_bond variable. >> >> BEST, >> -- afshin >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100628/90ce75e9/attachment.html From jonas.gauffin at gmail.com Mon Jun 28 12:24:58 2010 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Mon, 28 Jun 2010 21:24:58 +0200 Subject: [Freeswitch-users] Incompatible destination In-Reply-To: References: Message-ID: I don't That's why I asked u guys ;) On Mon, Jun 28, 2010 at 8:36 PM, David Ponzone wrote: > Note for later: never believe a provider when they say their packets are > valid :) > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 28/06/2010 ? 19:51, Jonas Gauffin a ?crit : > > Hello, > > I get this SDP from one endpoint: > > v=0 > o=- 40700591 0 IN IP4 130.244.X.XX > s=Cisco SDP 0 > c=IN IP4 130.244.X.XX > t=0 0 > m=audio 32512 RTP/AVP 8 0 99 102 101 > a=rtpmap:99 G.729a/8000 > a=rtpmap:102 G.729b/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > And this from the other end point: > > v=0 > o=- 33485646 0 IN IP4 130.244.Y.YY > s=Cisco SDP 0 > c=IN IP4 130.244.Y.YY > t=0 0 > m=audio 17142 RTP/AVP 8 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > FreeSwitch gives me INCOMPATIBLE_DESTINATION. My trunk provider says that > the SDP:s are valid. The first one looks OK by me. How about the second one? > > //Jonas > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100628/2c3882cb/attachment.html From jeroeng at thegreek.com Mon Jun 28 12:33:00 2010 From: jeroeng at thegreek.com (Jeroen C. van Gelderen) Date: Mon, 28 Jun 2010 14:33:00 -0500 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? In-Reply-To: <003e01cb16f2$09c7d920$1d578b60$@maly@molcs.org> References: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> <003e01cb16f2$09c7d920$1d578b60$@maly@molcs.org> Message-ID: <180EC6E8D9D94550A56D95A86BD3EDF2@mbnet.local> Hi Mark, I'm curious to see if you'd had any luck with DAHDI built-in echo cancellers such as MG2? (I figured since MG2 has been built into DAHDI for a while it would be slightly more likely to work than OSLEC. In my case neither MG2 nor OSLEC works.) In order to determine whether echo cancellation is in use with DAHDI you can check the output of lsdahdi: [root at elastix ~]# lsdahdi ### Span 1: XBUS-00/XPD-00 "Xorcom XPD #00/00: FXS" (MASTER) 1 FXS FXOKS (In use) (SWEC: OSLEC) (EC: OSLEC) 2 FXS FXOKS (In use) (SWEC: OSLEC) This shows FXS port 1 in off-hook with OSLEC in-use. FXS port 2 has OSLEC configured but is on-hook. If I were to run FreeSwitch the channel would be silent in this configuration. HTH, -Slim -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: jeroeng at thegreek.com Phone: +1 876 953 6182 x128 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Maly Sent: Monday, June 28, 2010 13:45 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? I'm attempting to use Freeswitch/DAHDHI and OSLEC, as well. I agree in commenting out the "echocancellation" line in zt.conf. The channels seem to work OK but the echo cancellation does not seem to work (I can see the kernel drivers for it, but I still have a great deal of echo). Mark -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen C. van Gelderen Sent: Saturday, June 26, 2010 7:57 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? Hi everybody, I have a feeling I must be missing the obvious... I've been trying to get echo canceling to work with Freeswitch/FreeTDM/DAHDI but I have been failing miserably for days. When I enable echo cancellation (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes silent. In absence of better suggestions I was hoping to replicate someone's "known working" configuration. Is anyone successfully running the following combination: - any x86 hardware - any recent flavor Linux - any recent version of DAHDI using FXO ports - FreeSwitch/FreeTDM trunk from git - OSLEC echo canceller (or MG2) - (BONUS:) Xorcom Astribank with FXO/FXS ports. I addition to much Googling I've tried most permutations of: - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix - x86 and x64 single and multicore CPUs. - Non-PAE kernels for 32-bit installs. - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) - MG2 and OSLEC echo cancellers - Freeswitch from Git trunk In each case everything configures fine to the point that Asterisk 1.6 will function with echo cancellation enabled. So we know that DAHDI layer works. But Freeswitch channels go silent when echo cancellation is enabled. Puzzled, -Slim -- Jeroen C. "Slim" van Gelderen _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 From david.ponzone at gmail.com Mon Jun 28 12:35:22 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 28 Jun 2010 21:35:22 +0200 Subject: [Freeswitch-users] Dialplan handling on call fails In-Reply-To: References: <96D730D0-57A6-4751-9194-1285B4DCC8D2@shaw.ca> Message-ID: <795F82FA-51D3-4232-875A-84D3AEDCEF2F@gmail.com> Roger, you're right, but I think there is a mix up here. In your dialplan, all your bridge actions will be executed because of the conditions they are included in, not because you use continue_on_fail. The common use of continue_on_fail looks like: David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 28/06/2010 ? 21:06, Roger Salloum a ?crit : > >From my understanding the break is used to handle what happens on > the evaluation of the condition. I have break=never as I want it to > try all rules for the extension even if the one above returns true > or false. > > ----- Original Message ----- > From: David Ponzone > Date: Monday, June 28, 2010 6:36 am > Subject: Re: [Freeswitch-users] Dialplan handling on call fails > To: freeswitch-users at lists.freeswitch.org > > > If you use continue_on_fail, I don't think you need to set > > break="never". > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > ?tablis > > ? l'intention exclusive de ses destinataires. Toute utilisation > > ou > > diffusion non autoris?e est interdite. Tout message ?lectronique > > est > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? > > au > > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. > > Si > > vous n'?tes pas destinataire de ce message, merci de le > > d?truire > > imm?diatement et d'avertir l'exp?diteur. > > > > > > > > > > Le 26/06/2010 ? 16:53, Roger Salloum a ?crit : > > > > > Hi, > > > > > > I'm trying to setup a dialplan such that if one particular > > route > > > fails it will try another. However, I do not want it to > > try another > > > route once it had recieved a 180/183 in response from a > > gateway. I > > > have not been able to determine how to accomplish this. > > > > > > For Example: > > > > > > > expression="^(1000123.*)$" > > > break="never"> > > > > application="set" data="hangup_after_bridge=true"/> > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > application="bridge" data="sofia/gateway/carrier1/$1"/> > > > > > > > > > > expression="^(1000.*)$" > > > break="never"> > > > > application="set" > > > data="hangup_after_bridge=RECOVER_ON_TIMER_EXPIRE"/> > > > > application="bridge" data="sofia/gateway/carrier2/$1"/> > > > > > > > > > > expression="^(1.*)$" > > > break="never"> > > > > application="set" data="hangup_after_bridge=true"/> > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > data="sofia/gateway/carrier3/$1"/>> > > > > > > So when 10001234567 is dialled i will match all 3. I'd like to > > be > > > able to try 1, if failed, try 2, if failed try 3. All calls go > > out > > > via an outbound proxy. > > > > > > Using the above examples if the gateway does not respond in > > time, > > > the proxy generates a 408 REQUEST TIMEOUT error message. It > > will > > > then fail out and try the next route. However, when the > > gateway > > > responds with a 180/183 but there is no answer after 2 minutes > > the > > > proxy, will generate a 480 NO ANSWER (also tried a 408 > > REQUEST > > > TIMEOUT ). When Freeswitch receives this message it fails, and > > then > > > attempts the third failure route. How do i prevent the > > dialplan from > > > continuing once it has received a 180/183 when no one answers > > the > > > phone? > > > > > > Thanks, > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.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/20100628/d85ef9c4/attachment-0001.html From david.ponzone at gmail.com Mon Jun 28 12:37:08 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Mon, 28 Jun 2010 21:37:08 +0200 Subject: [Freeswitch-users] SIP header on only one fork of a bridge In-Reply-To: <4C28EDB8.6080609@communicatefreely.net> References: <4C28EDB8.6080609@communicatefreely.net> Message-ID: <5FACB1C4-4BB1-4233-9354-02369E449748@gmail.com> use [] instead of {} in front of the destination David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 28/06/2010 ? 20:45, Tim St. Pierre a ?crit : > Hello list, > > I would like to bridge a call to multiple SIP endpoints, but add > different headers to each. > > I'm not entirely sure how to do this. I have no problem exporting a > SIP header that does what I > want for one destination, but I'm not sure how to set it for two. > > My application is that I want two IP phones to ring - one with the > internal ring-ring splash, the > others with a group-answer (single ring, then lamp flash only), for > administrative assistants, etc. > > How do I export different variables to each branch? > > Thanks! > > -Tim > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100628/07383d75/attachment.html From brian at freeswitch.org Mon Jun 28 12:42:24 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Jun 2010 14:42:24 -0500 Subject: [Freeswitch-users] SIP header on only one fork of a bridge In-Reply-To: <5FACB1C4-4BB1-4233-9354-02369E449748@gmail.com> References: <4C28EDB8.6080609@communicatefreely.net> <5FACB1C4-4BB1-4233-9354-02369E449748@gmail.com> Message-ID: <16BEF2FB-23BC-48F5-8C06-48E1BF66DDAB@freeswitch.org> You can use [] in front of each sofia/ channel. /b On Jun 28, 2010, at 2:37 PM, David Ponzone wrote: > use [] instead of {} in front of the destination From jeroeng at thegreek.com Mon Jun 28 13:01:42 2010 From: jeroeng at thegreek.com (Jeroen C. van Gelderen) Date: Mon, 28 Jun 2010 15:01:42 -0500 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? In-Reply-To: References: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> Message-ID: Hi Moises, Thank you. This is very helpful as it seems to detail the same problem. I tested with both OpenZap and FreeTDM yielding the same results. I think Anthony Minnessale is close when he writes: "Maybe there are some new ioctls or something they added for interop with OSLEC that we don't know about." Except that whatever changed in (the) DAHDI (API) seems to have broken all echo canceling (MG2 and OSLEC) somewhere between DAHDI and FreeTDM/OpenZap. Anthony also suggests: "If you still have the problem the next step would be to add some debugging code to the ozmod_zt read and write functions to see if it's sending data up to the app." I'm going to see if my C-fu is up to doing that. Anybody know where the debugging code should go for FreeTDM? Lastly, the only semi-relevant Google find I did was this [1]: "Make sure echotraining is disabled when using Oslec - this is not supported and if enabled will cause the channel to be silent (i.e. no audio will pass through)" In FreeTDM/OpenZap/FreeSwitch there doesn't seem to be an "echo training" knob to twiddle. Cheers, -Slim [1] http://www.rowetel.com/ucasterisk/oslec.html -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: jeroeng at thegreek.com Phone: +1 876 953 6182 x128 ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises Silva Sent: Monday, June 28, 2010 10:04 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? This may be of help: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055505.htm l Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Sat, Jun 26, 2010 at 8:57 PM, Jeroen C. van Gelderen wrote: Hi everybody, I have a feeling I must be missing the obvious... I've been trying to get echo canceling to work with Freeswitch/FreeTDM/DAHDI but I have been failing miserably for days. When I enable echo cancellation (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes silent. In absence of better suggestions I was hoping to replicate someone's "known working" configuration. Is anyone successfully running the following combination: - any x86 hardware - any recent flavor Linux - any recent version of DAHDI using FXO ports - FreeSwitch/FreeTDM trunk from git - OSLEC echo canceller (or MG2) - (BONUS:) Xorcom Astribank with FXO/FXS ports. I addition to much Googling I've tried most permutations of: - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix - x86 and x64 single and multicore CPUs. - Non-PAE kernels for 32-bit installs. - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) - MG2 and OSLEC echo cancellers - Freeswitch from Git trunk In each case everything configures fine to the point that Asterisk 1.6 will function with echo cancellation enabled. So we know that DAHDI layer works. But Freeswitch channels go silent when echo cancellation is enabled. Puzzled, -Slim -- Jeroen C. "Slim" van Gelderen _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 a.afzali2003 at gmail.com Mon Jun 28 14:03:36 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Tue, 29 Jun 2010 01:33:36 +0430 Subject: [Freeswitch-users] Using fifo_orbit_announce For On-hook Agent In-Reply-To: References: Message-ID: You right, I was going to use the fifo_orbit_announce to play personal greeting for each of agents that registered to just one FIFO. Then I've found the same old thread in this list which recommended to use bridge_pre_execute_aleg_app in dialstring for this purpose, so this well done. FYI my other thread about getting access to bridged session, also relates to this issue. With using of bridge_pre_execute_aleg_app variable, I launch a lua script which by getting uuid of other session plays personal greeting and some other things. This is the place I've asked about the preferred method of obtaining uuid of other session. appreciate -- afshin On Mon, Jun 28, 2010 at 9:54 PM, Michael Collins wrote: > I am not sure I understand what your setup is. Do you have a separate FIFO > queue for each agent? That would be the only way that each agent could have > his own specific announcement. Each FIFO queue has only one set of orbit > parameters (extension, context, dialplan, announcement file). What are you > trying to accomplish? > > -MC > > On Fri, Jun 25, 2010 at 6:39 AM, afshin afzali wrote: > >> Hi FreeSWITCH, >> >> I want to use fifo_orbit_announce to play specific agent greeting to his >> caller (can not insert it to caller dialplan). As my agents are on-hook >> (they use extensions to login / logout of queues) , I don't know if I could >> set this variable in login extension! If I can not use this way, is there >> any other way? >> >> appreciate all comments, >> BEST, >> -- afshin >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100629/23e5be24/attachment.html From a.afzali2003 at gmail.com Mon Jun 28 14:05:11 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Tue, 29 Jun 2010 01:35:11 +0430 Subject: [Freeswitch-users] How to Access To Bridged Seesion In-Reply-To: References: <60274886-04A5-4B8F-B5DE-BC120A6058F1@avgs.ca> Message-ID: Sure, I'll provide you the data. -- afshin On Mon, Jun 28, 2010 at 11:54 PM, Madovsky wrote: > Hi Mathieu, > > I tried to use Other-Leg-Unique-ID rather than signal_bond with this CLI > command > > fs_cli -x "expand uuid_send_dtmf ${uuid_getvar[callerID] > Other-Leg-Unique-ID)} 9 at 250" > > but doesn't work. signal_bond works well > > F > > ----- Original Message ----- > From: "Mathieu Rene" > To: > Sent: Sunday, June 27, 2010 9:27 PM > Subject: Re: [Freeswitch-users] How to Accesrs To Bridged Seesion > > > Hi, > > > > signal_bond will always work (thats whats being used internally). > However, if you are getting events from the core, you may have more luck > with Other-Leg-Unique-ID as the full list of channel variables isn't always > included. > > > > Cheers, > > > > Mathieu Rene > > Avant-Garde Solutions Inc > > Office: + 1 (514) 664-1044 x100 > > Cell: +1 (514) 664-1044 x200 > > mrene at avgs.ca > > > > > > > > > > On 2010-06-26, at 1:09 PM, afshin afzali wrote: > > > >> Hi FreeSWITCH, > >> > >> What is the preferred method for obtaining bridged session from the > current one? > >> I've just found the signal_bond variable. > >> > >> BEST, > >> -- afshin > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/635222e5/attachment-0001.html From roger_salloum at shaw.ca Mon Jun 28 14:44:08 2010 From: roger_salloum at shaw.ca (Roger Salloum) Date: Mon, 28 Jun 2010 14:44:08 -0700 Subject: [Freeswitch-users] Dialplan handling on call fails In-Reply-To: <795F82FA-51D3-4232-875A-84D3AEDCEF2F@gmail.com> References: <96D730D0-57A6-4751-9194-1285B4DCC8D2@shaw.ca> <795F82FA-51D3-4232-875A-84D3AEDCEF2F@gmail.com> Message-ID: Hi David, Thanks for clearing that up. I tried when the call establishes and it would hang up, which is controlled by hangup_after_bridge. Do you have any suggestions for how i would get the failover without requiring all the bridges to be within the same condition? So that i can get failover to a less best match? ----- Original Message ----- From: David Ponzone Date: Monday, June 28, 2010 12:36 pm Subject: Re: [Freeswitch-users] Dialplan handling on call fails To: freeswitch-users at lists.freeswitch.org > Roger, > > you're right, but I think there is a mix up here. > In your dialplan, all your bridge actions will be executed > because of? > the conditions they are included in, not because you use? > continue_on_fail. > > The common use of continue_on_fail looks like: > > > > > > > > David Ponzone? Direction Technique > email: david.ponzone at ipeva.fr > tel:????? 01 74 03 18 97 > gsm:?? 06 66 98 76 34 > > Service Client IPeva > tel:????? 0811 46 26 26 > www.ipeva.fr? -?? www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis? > ? l'intention exclusive de ses destinataires. Toute utilisation > ou? > diffusion non autoris?e est interdite. Tout message ?lectronique > est? > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > titre? > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > n'?tes? > pas destinataire de ce message, merci de le d?truire > imm?diatement et? > d'avertir l'exp?diteur. > > > > > David Ponzone? Direction Technique > email: david.ponzone at ipeva.fr > tel:????? 01 74 03 18 97 > gsm:?? 06 66 98 76 34 > > Service Client IPeva > tel:????? 0811 46 26 26 > www.ipeva.fr? -?? www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis? > ? l'intention exclusive de ses destinataires. Toute utilisation > ou? > diffusion non autoris?e est interdite. Tout message ?lectronique > est? > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > titre? > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > n'?tes? > pas destinataire de ce message, merci de le d?truire > imm?diatement et? > d'avertir l'exp?diteur. > > > > > Le 28/06/2010 ? 21:06, Roger Salloum a ?crit : > > > >From my understanding the break is used to handle what > happens on? > > the evaluation of the condition. I have break=never as I want > it to? > > try all rules for the extension even if the one above returns > true? > > or false. > > > > ----- Original Message ----- > > From: David Ponzone > > Date: Monday, June 28, 2010 6:36 am > > Subject: Re: [Freeswitch-users] Dialplan handling on call fails > > To: freeswitch-users at lists.freeswitch.org > > > > > If you use continue_on_fail, I don't think you need to set > > > break="never". > > > > > > David Ponzone? Direction Technique > > > email: david.ponzone at ipeva.fr > > > tel:????? 01 74 03 18 97 > > > gsm:?? 06 66 98 76 34 > > > > > > Service Client IPeva > > > tel:????? 0811 46 26 26 > > > www.ipeva.fr? -?? www.ipeva-studio.com > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > ?tablis > > > ? l'intention exclusive de ses destinataires. Toute utilisation > > > ou > > > diffusion non autoris?e est interdite. Tout message ?lectronique > > > est > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? > > > au > > > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. > > > Si > > > vous n'?tes pas destinataire de ce message, merci de le > > > d?truire > > > imm?diatement et d'avertir l'exp?diteur. > > > > > > > > > > > > > > > Le 26/06/2010 ? 16:53, Roger Salloum a ?crit : > > > > > > > Hi, > > > > > > > > I'm trying to setup a dialplan such that if one particular > > > route > > > > fails it will try another.? However, I do not want it to > > > try another > > > > route once it had recieved a 180/183 in response from a > > > gateway. I > > > > have not been able to determine how to accomplish this. > > > > > > > > For Example: > > > > > > > >?? > > expression="^(1000123.*)$" > > > > break="never"> > > > >?????? > > application="set" data="hangup_after_bridge=true"/> > > > >? > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > >?????? > > application="bridge" data="sofia/gateway/carrier1/$1"/> > > > >?? > > > > > > > >?? > > expression="^(1000.*)$" > > > > break="never"> > > > >?????? > > application="set" > > > > data="hangup_after_bridge=RECOVER_ON_TIMER_EXPIRE"/> > > > >?????? > > application="bridge" data="sofia/gateway/carrier2/$1"/> > > > >?? > > > > > > > >?? > > expression="^(1.*)$" > > > > break="never"> > > > >?????? > > application="set" data="hangup_after_bridge=true"/> > > > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > >???? > > data="sofia/gateway/carrier3/$1"/>>?? > > > > > > > > So when 10001234567 is dialled i will match all 3. I'd > like to > > > be > > > > able to try 1, if failed, try 2, if failed try 3. All > calls go > > > out > > > > via an outbound proxy. > > > > > > > > Using the above examples if the gateway does not respond in > > > time, > > > > the proxy generates a 408 REQUEST TIMEOUT error message. It > > > will > > > > then fail out and try the next route. However, when the > > > gateway > > > > responds with a 180/183 but there is no answer after 2 minutes > > > the > > > > proxy, will generate a 480 NO ANSWER (also tried a 408 > > > REQUEST > > > > TIMEOUT ). When Freeswitch receives this message it fails, and > > > then > > > > attempts the third failure route. How do i prevent the > > > dialplan from > > > > continuing once it has received a 180/183 when no one answers > > > the > > > > phone? > > > > > > > > Thanks, > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.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/20100628/9f15f8d1/attachment.html From david.ponzone at gmail.com Mon Jun 28 15:43:55 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Tue, 29 Jun 2010 00:43:55 +0200 Subject: [Freeswitch-users] Dialplan handling on call fails In-Reply-To: References: <96D730D0-57A6-4751-9194-1285B4DCC8D2@shaw.ca> <795F82FA-51D3-4232-875A-84D3AEDCEF2F@gmail.com> Message-ID: <5D032079-D1B3-4B5B-8DA7-059FB8549D53@gmail.com> Perhaps by using continue_on_fail and putting a transfer just after the bridge, so you can jump to another extension/dialplan ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 28/06/2010 ? 23:44, Roger Salloum a ?crit : > Hi David, > > Thanks for clearing that up. I tried when the call establishes and > it would hang up, which is controlled by hangup_after_bridge. Do you > have any suggestions for how i would get the failover without > requiring all the bridges to be within the same condition? So that i > can get failover to a less best match? > > ----- Original Message ----- > From: David Ponzone > Date: Monday, June 28, 2010 12:36 pm > Subject: Re: [Freeswitch-users] Dialplan handling on call fails > To: freeswitch-users at lists.freeswitch.org > > > Roger, > > > > you're right, but I think there is a mix up here. > > In your dialplan, all your bridge actions will be executed > > because of > > the conditions they are included in, not because you use > > continue_on_fail. > > > > The common use of continue_on_fail looks like: > > > > > > > > > > > > > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > ?tablis > > ? l'intention exclusive de ses destinataires. Toute utilisation > > ou > > diffusion non autoris?e est interdite. Tout message ?lectronique > > est > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > titre > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > n'?tes > > pas destinataire de ce message, merci de le d?truire > > imm?diatement et > > d'avertir l'exp?diteur. > > > > > > > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > ?tablis > > ? l'intention exclusive de ses destinataires. Toute utilisation > > ou > > diffusion non autoris?e est interdite. Tout message ?lectronique > > est > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > titre > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > n'?tes > > pas destinataire de ce message, merci de le d?truire > > imm?diatement et > > d'avertir l'exp?diteur. > > > > > > > > > > Le 28/06/2010 ? 21:06, Roger Salloum a ?crit : > > > > > >From my understanding the break is used to handle what > > happens on > > > the evaluation of the condition. I have break=never as I want > > it to > > > try all rules for the extension even if the one above returns > > true > > > or false. > > > > > > ----- Original Message ----- > > > From: David Ponzone > > > Date: Monday, June 28, 2010 6:36 am > > > Subject: Re: [Freeswitch-users] Dialplan handling on call fails > > > To: freeswitch-users at lists.freeswitch.org > > > > > > > If you use continue_on_fail, I don't think you need to set > > > > break="never". > > > > > > > > David Ponzone Direction Technique > > > > email: david.ponzone at ipeva.fr > > > > tel: 01 74 03 18 97 > > > > gsm: 06 66 98 76 34 > > > > > > > > Service Client IPeva > > > > tel: 0811 46 26 26 > > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > > ?tablis > > > > ? l'intention exclusive de ses destinataires. Toute utilisation > > > > ou > > > > diffusion non autoris?e est interdite. Tout message ?lectronique > > > > est > > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? > > > > au > > > > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. > > > > Si > > > > vous n'?tes pas destinataire de ce message, merci de le > > > > d?truire > > > > imm?diatement et d'avertir l'exp?diteur. > > > > > > > > > > > > > > > > > > > > Le 26/06/2010 ? 16:53, Roger Salloum a ?crit : > > > > > > > > > Hi, > > > > > > > > > > I'm trying to setup a dialplan such that if one particular > > > > route > > > > > fails it will try another. However, I do not want it to > > > > try another > > > > > route once it had recieved a 180/183 in response from a > > > > gateway. I > > > > > have not been able to determine how to accomplish this. > > > > > > > > > > For Example: > > > > > > > > > > > > > expression="^(1000123.*)$" > > > > > break="never"> > > > > > > > > application="set" data="hangup_after_bridge=true"/> > > > > > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > > > > > application="bridge" data="sofia/gateway/carrier1/$1"/> > > > > > > > > > > > > > > > > > > expression="^(1000.*)$" > > > > > break="never"> > > > > > > > > application="set" > > > > > data="hangup_after_bridge=RECOVER_ON_TIMER_EXPIRE"/> > > > > > > > > application="bridge" data="sofia/gateway/carrier2/$1"/> > > > > > > > > > > > > > > > > > > expression="^(1.*)$" > > > > > break="never"> > > > > > > > > application="set" data="hangup_after_bridge=true"/> > > > > > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > > > > > data="sofia/gateway/carrier3/$1"/>> > > > > > > > > > > So when 10001234567 is dialled i will match all 3. I'd > > like to > > > > be > > > > > able to try 1, if failed, try 2, if failed try 3. All > > calls go > > > > out > > > > > via an outbound proxy. > > > > > > > > > > Using the above examples if the gateway does not respond in > > > > time, > > > > > the proxy generates a 408 REQUEST TIMEOUT error message. It > > > > will > > > > > then fail out and try the next route. However, when the > > > > gateway > > > > > responds with a 180/183 but there is no answer after 2 minutes > > > > the > > > > > proxy, will generate a 480 NO ANSWER (also tried a 408 > > > > REQUEST > > > > > TIMEOUT ). When Freeswitch receives this message it fails, and > > > > then > > > > > attempts the third failure route. How do i prevent the > > > > dialplan from > > > > > continuing once it has received a 180/183 when no one answers > > > > the > > > > > phone? > > > > > > > > > > Thanks, > > > > > > > > > > _______________________________________________ > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.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 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/3fcb03d2/attachment-0001.html From moises.silva at gmail.com Mon Jun 28 15:46:27 2010 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 28 Jun 2010 18:46:27 -0400 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? In-Reply-To: References: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> Message-ID: That is what echo_train_level in zt.conf is used for. I wonder if we should not be calling the echo train ioctl when is 0, may be the sole fact of calling the echo train ioctl is screwing things up. Try going to src/ftmod/ftmod_zt/ftmod_zt.c around line 642 and comment the if (ioctl(ftdmchan->sockfd, codes.ECHOTRAIN, &len)) { } the whole if block should be commented, you can use #if 0 #endif, or regular C comments. I took a look at the dahdi drivers and it seems even when this setting is 0, then set the state to ECHO_MODE_PRETRAINING, and that can be what OSLEC does not like at all. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Mon, Jun 28, 2010 at 4:01 PM, Jeroen C. van Gelderen < jeroeng at thegreek.com> wrote: > Hi Moises, > > Thank you. This is very helpful as it seems to detail the same problem. I > tested with both OpenZap and FreeTDM yielding the same results. > > I think Anthony Minnessale is close when he writes: > > "Maybe there are some new ioctls or something > they added for interop with OSLEC that we don't > know about." > > Except that whatever changed in (the) DAHDI (API) seems to have broken all > echo canceling (MG2 and OSLEC) somewhere between DAHDI and FreeTDM/OpenZap. > > Anthony also suggests: > > "If you still have the problem the next step > would be to add some debugging code to the > ozmod_zt read and write functions to see if > it's sending data up to the app." > > I'm going to see if my C-fu is up to doing that. Anybody know where the > debugging code should go for FreeTDM? > > Lastly, the only semi-relevant Google find I did was this [1]: > > "Make sure echotraining is disabled when using > Oslec - this is not supported and if enabled > will cause the channel to be silent (i.e. no > audio will pass through)" > > In FreeTDM/OpenZap/FreeSwitch there doesn't seem to be an "echo training" > knob to twiddle. > > Cheers, > -Slim > > [1] http://www.rowetel.com/ucasterisk/oslec.html > > -- > Jeroen C. "Slim" van Gelderen > Olympic Sports Data Services > Email: jeroeng at thegreek.com > Phone: +1 876 953 6182 x128 > ________________________________________ > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises > Silva > Sent: Monday, June 28, 2010 10:04 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? > > This may be of help: > > > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055505.htm > l > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 > Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > On Sat, Jun 26, 2010 at 8:57 PM, Jeroen C. van Gelderen > wrote: > Hi everybody, > > I have a feeling I must be missing the obvious... > > I've been trying to get echo canceling to work with > Freeswitch/FreeTDM/DAHDI > but I have been failing miserably for days. When I enable echo cancellation > (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes > silent. > > In absence of better suggestions I was hoping to replicate someone's "known > working" configuration. Is anyone successfully running the following > combination: > > - any x86 hardware > - any recent flavor Linux > - any recent version of DAHDI using FXO ports > - FreeSwitch/FreeTDM trunk from git > - OSLEC echo canceller (or MG2) > - (BONUS:) Xorcom Astribank with FXO/FXS ports. > > I addition to much Googling I've tried most permutations of: > - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix > - x86 and x64 single and multicore CPUs. > - Non-PAE kernels for 32-bit installs. > - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) > - MG2 and OSLEC echo cancellers > - Freeswitch from Git trunk > > In each case everything configures fine to the point that Asterisk 1.6 will > function with echo cancellation enabled. So we know that DAHDI layer works. > But Freeswitch channels go silent when echo cancellation is enabled. > > Puzzled, > -Slim > > -- > Jeroen C. "Slim" van Gelderen > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100628/f4f167e4/attachment.html From jeroeng at thegreek.com Mon Jun 28 17:19:37 2010 From: jeroeng at thegreek.com (Jeroen C. van Gelderen) Date: Mon, 28 Jun 2010 19:19:37 -0500 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? In-Reply-To: References: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> Message-ID: <844789406B5046C7A02F94585161093C@mbnet.local> Hi Moises, I basically arrived at the same conclusion about echo training ioctl and I commented out the ECHOTRAIN ioctl in FreeTDM. FreeSwitch does indeed pass audio if the ECHOTRAIN ioctl does NOT get called.. Cheers, -Slim -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: jeroeng at thegreek.com Phone: +1 876 953 6182 x128 _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises Silva Sent: Monday, June 28, 2010 17:46 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? That is what echo_train_level in zt.conf is used for. I wonder if we should not be calling the echo train ioctl when is 0, may be the sole fact of calling the echo train ioctl is screwing things up. Try going to src/ftmod/ftmod_zt/ftmod_zt.c around line 642 and comment the if (ioctl(ftdmchan->sockfd, codes.ECHOTRAIN, &len)) { } the whole if block should be commented, you can use #if 0 #endif, or regular C comments. I took a look at the dahdi drivers and it seems even when this setting is 0, then set the state to ECHO_MODE_PRETRAINING, and that can be what OSLEC does not like at all. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t.. 1 905 474 1990 x 128 | e. moy at sangoma.com On Mon, Jun 28, 2010 at 4:01 PM, Jeroen C. van Gelderen wrote: Hi Moises, Thank you. This is very helpful as it seems to detail the same problem. I tested with both OpenZap and FreeTDM yielding the same results. I think Anthony Minnessale is close when he writes: "Maybe there are some new ioctls or something they added for interop with OSLEC that we don't know about." Except that whatever changed in (the) DAHDI (API) seems to have broken all echo canceling (MG2 and OSLEC) somewhere between DAHDI and FreeTDM/OpenZap. Anthony also suggests: "If you still have the problem the next step would be to add some debugging code to the ozmod_zt read and write functions to see if it's sending data up to the app." I'm going to see if my C-fu is up to doing that. Anybody know where the debugging code should go for FreeTDM? Lastly, the only semi-relevant Google find I did was this [1]: "Make sure echotraining is disabled when using Oslec - this is not supported and if enabled will cause the channel to be silent (i.e. no audio will pass through)" In FreeTDM/OpenZap/FreeSwitch there doesn't seem to be an "echo training" knob to twiddle. Cheers, -Slim [1] http://www.rowetel.com/ucasterisk/oslec.html -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: jeroeng at thegreek.com Phone: +1 876 953 6182 x128 ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises Silva Sent: Monday, June 28, 2010 10:04 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? This may be of help: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055505.htm l Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma..com On Sat, Jun 26, 2010 at 8:57 PM, Jeroen C. van Gelderen wrote: Hi everybody, I have a feeling I must be missing the obvious... I've been trying to get echo canceling to work with Freeswitch/FreeTDM/DAHDI but I have been failing miserably for days. When I enable echo cancellation (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes silent. In absence of better suggestions I was hoping to replicate someone's "known working" configuration. Is anyone successfully running the following combination: - any x86 hardware - any recent flavor Linux - any recent version of DAHDI using FXO ports - FreeSwitch/FreeTDM trunk from git - OSLEC echo canceller (or MG2) - (BONUS:) Xorcom Astribank with FXO/FXS ports. I addition to much Googling I've tried most permutations of: - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix - x86 and x64 single and multicore CPUs. - Non-PAE kernels for 32-bit installs. - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) - MG2 and OSLEC echo cancellers - Freeswitch from Git trunk In each case everything configures fine to the point that Asterisk 1.6 will function with echo cancellation enabled. So we know that DAHDI layer works. But Freeswitch channels go silent when echo cancellation is enabled. Puzzled, -Slim -- Jeroen C. "Slim" van Gelderen _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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/20100628/9be256ce/attachment-0001.html From jeroeng at thegreek.com Mon Jun 28 17:51:30 2010 From: jeroeng at thegreek.com (Jeroen C. van Gelderen) Date: Mon, 28 Jun 2010 19:51:30 -0500 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? In-Reply-To: <844789406B5046C7A02F94585161093C@mbnet.local> References: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> <844789406B5046C7A02F94585161093C@mbnet.local> Message-ID: Hi Moises, I can confirm that the following patch solves the problem: [root at elastix freeswitch]# git diff diff --git a/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c b/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c index a2eacac..b4a3acf 100644 --- a/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c +++ b/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c @@ -639,8 +639,9 @@ static FIO_OPEN_FUNCTION(zt_open) } if (ioctl(ftdmchan->sockfd, codes.ECHOCANCEL, &len)) { ftdm_log(FTDM_LOG_WARNING, "Echo cancel not available for %d:%d\n", ftdmchan->span_id, ftdmchan->chan_id); - } else if (zt_globals.etlevel >= 0) { + } else if (zt_globals.etlevel > 0) { len = zt_globals.etlevel; + ftdm_log(FTDM_LOG_INFO, "Enabling echo training %d:%d\n", ftdmchan->span_id, ftdmchan->chan_id); if (ioctl(ftdmchan->sockfd, codes.ECHOTRAIN, &len)) { ftdm_log(FTDM_LOG_WARNING, "Echo training not available for %d:%d\n", ftdmchan->span_id, ftdmchan->chan_id); } [root at elastix freeswitch]# Cheers, -Slim -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: jeroeng at thegreek.com Phone: +1 876 953 6182 x128 _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen C. van Gelderen Sent: Monday, June 28, 2010 19:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? Hi Moises, I basically arrived at the same conclusion about echo training ioctl and I commented out the ECHOTRAIN ioctl in FreeTDM. FreeSwitch does indeed pass audio if the ECHOTRAIN ioctl does NOT get called.. Cheers, -Slim -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: jeroeng at thegreek.com Phone: +1 876 953 6182 x128 _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises Silva Sent: Monday, June 28, 2010 17:46 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? That is what echo_train_level in zt.conf is used for. I wonder if we should not be calling the echo train ioctl when is 0, may be the sole fact of calling the echo train ioctl is screwing things up. Try going to src/ftmod/ftmod_zt/ftmod_zt.c around line 642 and comment the if (ioctl(ftdmchan->sockfd, codes.ECHOTRAIN, &len)) { } the whole if block should be commented, you can use #if 0 #endif, or regular C comments. I took a look at the dahdi drivers and it seems even when this setting is 0, then set the state to ECHO_MODE_PRETRAINING, and that can be what OSLEC does not like at all. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t.. 1 905 474 1990 x 128 | e. moy at sangoma.com On Mon, Jun 28, 2010 at 4:01 PM, Jeroen C. van Gelderen wrote: Hi Moises, Thank you. This is very helpful as it seems to detail the same problem. I tested with both OpenZap and FreeTDM yielding the same results. I think Anthony Minnessale is close when he writes: "Maybe there are some new ioctls or something they added for interop with OSLEC that we don't know about." Except that whatever changed in (the) DAHDI (API) seems to have broken all echo canceling (MG2 and OSLEC) somewhere between DAHDI and FreeTDM/OpenZap. Anthony also suggests: "If you still have the problem the next step would be to add some debugging code to the ozmod_zt read and write functions to see if it's sending data up to the app." I'm going to see if my C-fu is up to doing that. Anybody know where the debugging code should go for FreeTDM? Lastly, the only semi-relevant Google find I did was this [1]: "Make sure echotraining is disabled when using Oslec - this is not supported and if enabled will cause the channel to be silent (i.e. no audio will pass through)" In FreeTDM/OpenZap/FreeSwitch there doesn't seem to be an "echo training" knob to twiddle. Cheers, -Slim [1] http://www.rowetel.com/ucasterisk/oslec.html -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: jeroeng at thegreek.com Phone: +1 876 953 6182 x128 ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises Silva Sent: Monday, June 28, 2010 10:04 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? This may be of help: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055505.htm l Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma..com On Sat, Jun 26, 2010 at 8:57 PM, Jeroen C. van Gelderen wrote: Hi everybody, I have a feeling I must be missing the obvious... I've been trying to get echo canceling to work with Freeswitch/FreeTDM/DAHDI but I have been failing miserably for days. When I enable echo cancellation (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes silent. In absence of better suggestions I was hoping to replicate someone's "known working" configuration. Is anyone successfully running the following combination: - any x86 hardware - any recent flavor Linux - any recent version of DAHDI using FXO ports - FreeSwitch/FreeTDM trunk from git - OSLEC echo canceller (or MG2) - (BONUS:) Xorcom Astribank with FXO/FXS ports. I addition to much Googling I've tried most permutations of: - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix - x86 and x64 single and multicore CPUs. - Non-PAE kernels for 32-bit installs. - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) - MG2 and OSLEC echo cancellers - Freeswitch from Git trunk In each case everything configures fine to the point that Asterisk 1.6 will function with echo cancellation enabled. So we know that DAHDI layer works. But Freeswitch channels go silent when echo cancellation is enabled. Puzzled, -Slim -- Jeroen C. "Slim" van Gelderen _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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/20100628/840e7931/attachment.html From moises.silva at gmail.com Mon Jun 28 21:18:11 2010 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 29 Jun 2010 00:18:11 -0400 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? In-Reply-To: References: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> <844789406B5046C7A02F94585161093C@mbnet.local> Message-ID: Thanks for checking, I just committed the fix to freetdm. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Mon, Jun 28, 2010 at 8:51 PM, Jeroen C. van Gelderen < jeroeng at thegreek.com> wrote: > Hi Moises, > > > > I can confirm that the following patch solves the problem: > > > > [root at elastix freeswitch]# git diff > > diff --git a/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c > b/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c > > index a2eacac..b4a3acf 100644 > > --- a/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c > > +++ b/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c > > @@ -639,8 +639,9 @@ static FIO_OPEN_FUNCTION(zt_open) > > } > > if (ioctl(ftdmchan->sockfd, codes.ECHOCANCEL, > &len)) { > > ftdm_log(FTDM_LOG_WARNING, "Echo cancel not > available for %d:%d\n", ftdmchan->span_id, ftdmchan->chan_id); > > - } else if (zt_globals.etlevel >= 0) { > > + } else if (zt_globals.etlevel > 0) { > > len = zt_globals.etlevel; > > + ftdm_log(FTDM_LOG_INFO, "Enabling echo > training %d:%d\n", ftdmchan->span_id, ftdmchan->chan_id); > > if (ioctl(ftdmchan->sockfd, > codes.ECHOTRAIN, &len)) { > > ftdm_log(FTDM_LOG_WARNING, "Echo > training not available for %d:%d\n", ftdmchan->span_id, ftdmchan->chan_id); > > } > > [root at elastix freeswitch]# > > > > Cheers, > -Slim > -- > Jeroen C. "Slim" van Gelderen > Olympic Sports Data Services > Email: jeroeng at thegreek.com > Phone: +1 876 953 6182 x128 > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jeroen C. > van Gelderen > *Sent:* Monday, June 28, 2010 19:20 > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? > > > > Hi Moises, > > > > I basically arrived at the same conclusion about echo training ioctl and I > commented out the ECHOTRAIN ioctl in FreeTDM. FreeSwitch does indeed pass > audio if the ECHOTRAIN ioctl does NOT get called.. > > Cheers, > -Slim > -- > Jeroen C. "Slim" van Gelderen > Olympic Sports Data Services > Email: jeroeng at thegreek.com > Phone: +1 876 953 6182 x128 > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Moises Silva > *Sent:* Monday, June 28, 2010 17:46 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? > > > > That is what echo_train_level in zt.conf is used for. I wonder if we should > not be calling the echo train ioctl when is 0, may be the sole fact of > calling the echo train ioctl is screwing things up. > > Try going to src/ftmod/ftmod_zt/ftmod_zt.c around line 642 and comment the > > if (ioctl(ftdmchan->sockfd, codes.ECHOTRAIN, &len)) { > } > > the whole if block should be commented, you can use #if 0 #endif, or > regular C comments. > > I took a look at the dahdi drivers and it seems even when this setting is > 0, then set the state to ECHO_MODE_PRETRAINING, and that can be what OSLEC > does not like at all. > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t.. 1 905 474 1990 x 128 | e. moy at sangoma.com > > On Mon, Jun 28, 2010 at 4:01 PM, Jeroen C. van Gelderen < > jeroeng at thegreek.com> wrote: > > Hi Moises, > > Thank you. This is very helpful as it seems to detail the same problem. I > tested with both OpenZap and FreeTDM yielding the same results. > > I think Anthony Minnessale is close when he writes: > > "Maybe there are some new ioctls or something > they added for interop with OSLEC that we don't > know about." > > Except that whatever changed in (the) DAHDI (API) seems to have broken all > echo canceling (MG2 and OSLEC) somewhere between DAHDI and FreeTDM/OpenZap. > > Anthony also suggests: > > "If you still have the problem the next step > would be to add some debugging code to the > ozmod_zt read and write functions to see if > it's sending data up to the app." > > I'm going to see if my C-fu is up to doing that. Anybody know where the > debugging code should go for FreeTDM? > > Lastly, the only semi-relevant Google find I did was this [1]: > > "Make sure echotraining is disabled when using > Oslec - this is not supported and if enabled > will cause the channel to be silent (i.e. no > audio will pass through)" > > In FreeTDM/OpenZap/FreeSwitch there doesn't seem to be an "echo training" > knob to twiddle. > > Cheers, > -Slim > > [1] http://www.rowetel.com/ucasterisk/oslec.html > > > -- > Jeroen C. "Slim" van Gelderen > Olympic Sports Data Services > Email: jeroeng at thegreek.com > Phone: +1 876 953 6182 x128 > > ________________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises > Silva > Sent: Monday, June 28, 2010 10:04 > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? > > This may be of help: > > > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055505.htm > l > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 > Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma..com > > > On Sat, Jun 26, 2010 at 8:57 PM, Jeroen C. van Gelderen > wrote: > Hi everybody, > > I have a feeling I must be missing the obvious... > > I've been trying to get echo canceling to work with > Freeswitch/FreeTDM/DAHDI > but I have been failing miserably for days. When I enable echo cancellation > (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes > silent. > > In absence of better suggestions I was hoping to replicate someone's "known > working" configuration. Is anyone successfully running the following > combination: > > - any x86 hardware > - any recent flavor Linux > - any recent version of DAHDI using FXO ports > - FreeSwitch/FreeTDM trunk from git > - OSLEC echo canceller (or MG2) > - (BONUS:) Xorcom Astribank with FXO/FXS ports. > > I addition to much Googling I've tried most permutations of: > - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix > - x86 and x64 single and multicore CPUs. > - Non-PAE kernels for 32-bit installs. > - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) > - MG2 and OSLEC echo cancellers > - Freeswitch from Git trunk > > In each case everything configures fine to the point that Asterisk 1.6 will > function with echo cancellation enabled. So we know that DAHDI layer works. > But Freeswitch channels go silent when echo cancellation is enabled. > > Puzzled, > -Slim > > -- > Jeroen C. "Slim" van Gelderen > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/31549c58/attachment-0001.html From tony.tin at noahmedia.com.hk Mon Jun 28 23:29:48 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Tue, 29 Jun 2010 14:29:48 +0800 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? In-Reply-To: References: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> <844789406B5046C7A02F94585161093C@mbnet.local> Message-ID: Hi, I'm using openzap and libpri. I patched the "ozmod_zt.c" in the same way and the same problem which is also applied to openzap is fixed. Will anyone update the openzap source ? Regards, Tony On Tue, Jun 29, 2010 at 8:51 AM, Jeroen C. van Gelderen < jeroeng at thegreek.com> wrote: > Hi Moises, > > > > I can confirm that the following patch solves the problem: > > > > [root at elastix freeswitch]# git diff > > diff --git a/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c > b/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c > > index a2eacac..b4a3acf 100644 > > --- a/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c > > +++ b/libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c > > @@ -639,8 +639,9 @@ static FIO_OPEN_FUNCTION(zt_open) > > } > > if (ioctl(ftdmchan->sockfd, codes.ECHOCANCEL, > &len)) { > > ftdm_log(FTDM_LOG_WARNING, "Echo cancel not > available for %d:%d\n", ftdmchan->span_id, ftdmchan->chan_id); > > - } else if (zt_globals.etlevel >= 0) { > > + } else if (zt_globals.etlevel > 0) { > > len = zt_globals.etlevel; > > + ftdm_log(FTDM_LOG_INFO, "Enabling echo > training %d:%d\n", ftdmchan->span_id, ftdmchan->chan_id); > > if (ioctl(ftdmchan->sockfd, > codes.ECHOTRAIN, &len)) { > > ftdm_log(FTDM_LOG_WARNING, "Echo > training not available for %d:%d\n", ftdmchan->span_id, ftdmchan->chan_id); > > } > > [root at elastix freeswitch]# > > > > Cheers, > -Slim > -- > Jeroen C. "Slim" van Gelderen > Olympic Sports Data Services > Email: jeroeng at thegreek.com > Phone: +1 876 953 6182 x128 > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jeroen C. > van Gelderen > *Sent:* Monday, June 28, 2010 19:20 > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? > > > > Hi Moises, > > > > I basically arrived at the same conclusion about echo training ioctl and I > commented out the ECHOTRAIN ioctl in FreeTDM. FreeSwitch does indeed pass > audio if the ECHOTRAIN ioctl does NOT get called.. > > Cheers, > -Slim > -- > Jeroen C. "Slim" van Gelderen > Olympic Sports Data Services > Email: jeroeng at thegreek.com > Phone: +1 876 953 6182 x128 > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Moises Silva > *Sent:* Monday, June 28, 2010 17:46 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? > > > > That is what echo_train_level in zt.conf is used for. I wonder if we should > not be calling the echo train ioctl when is 0, may be the sole fact of > calling the echo train ioctl is screwing things up. > > Try going to src/ftmod/ftmod_zt/ftmod_zt.c around line 642 and comment the > > if (ioctl(ftdmchan->sockfd, codes.ECHOTRAIN, &len)) { > } > > the whole if block should be commented, you can use #if 0 #endif, or > regular C comments. > > I took a look at the dahdi drivers and it seems even when this setting is > 0, then set the state to ECHO_MODE_PRETRAINING, and that can be what OSLEC > does not like at all. > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t.. 1 905 474 1990 x 128 | e. moy at sangoma.com > > On Mon, Jun 28, 2010 at 4:01 PM, Jeroen C. van Gelderen < > jeroeng at thegreek.com> wrote: > > Hi Moises, > > Thank you. This is very helpful as it seems to detail the same problem. I > tested with both OpenZap and FreeTDM yielding the same results. > > I think Anthony Minnessale is close when he writes: > > "Maybe there are some new ioctls or something > they added for interop with OSLEC that we don't > know about." > > Except that whatever changed in (the) DAHDI (API) seems to have broken all > echo canceling (MG2 and OSLEC) somewhere between DAHDI and FreeTDM/OpenZap. > > Anthony also suggests: > > "If you still have the problem the next step > would be to add some debugging code to the > ozmod_zt read and write functions to see if > it's sending data up to the app." > > I'm going to see if my C-fu is up to doing that. Anybody know where the > debugging code should go for FreeTDM? > > Lastly, the only semi-relevant Google find I did was this [1]: > > "Make sure echotraining is disabled when using > Oslec - this is not supported and if enabled > will cause the channel to be silent (i.e. no > audio will pass through)" > > In FreeTDM/OpenZap/FreeSwitch there doesn't seem to be an "echo training" > knob to twiddle. > > Cheers, > -Slim > > [1] http://www.rowetel.com/ucasterisk/oslec.html > > > -- > Jeroen C. "Slim" van Gelderen > Olympic Sports Data Services > Email: jeroeng at thegreek.com > Phone: +1 876 953 6182 x128 > > ________________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises > Silva > Sent: Monday, June 28, 2010 10:04 > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? > > This may be of help: > > > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055505.htm > l > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 > Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma..com > > > On Sat, Jun 26, 2010 at 8:57 PM, Jeroen C. van Gelderen > wrote: > Hi everybody, > > I have a feeling I must be missing the obvious... > > I've been trying to get echo canceling to work with > Freeswitch/FreeTDM/DAHDI > but I have been failing miserably for days. When I enable echo cancellation > (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes > silent. > > In absence of better suggestions I was hoping to replicate someone's "known > working" configuration. Is anyone successfully running the following > combination: > > - any x86 hardware > - any recent flavor Linux > - any recent version of DAHDI using FXO ports > - FreeSwitch/FreeTDM trunk from git > - OSLEC echo canceller (or MG2) > - (BONUS:) Xorcom Astribank with FXO/FXS ports. > > I addition to much Googling I've tried most permutations of: > - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix > - x86 and x64 single and multicore CPUs. > - Non-PAE kernels for 32-bit installs. > - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) > - MG2 and OSLEC echo cancellers > - Freeswitch from Git trunk > > In each case everything configures fine to the point that Asterisk 1.6 will > function with echo cancellation enabled. So we know that DAHDI layer works. > But Freeswitch channels go silent when echo cancellation is enabled. > > Puzzled, > -Slim > > -- > Jeroen C. "Slim" van Gelderen > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/2f4f0235/attachment.html From mike at jerris.com Tue Jun 29 00:13:25 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 29 Jun 2010 03:13:25 -0400 Subject: [Freeswitch-users] ESL: Question about BACKGROUND_JOB events In-Reply-To: <4C22301C.3080405@ewetel.de> References: <4C22301C.3080405@ewetel.de> Message-ID: <4274374276864666200@unknownmsgid> origination_uuid On Jun 23, 2010, at 12:02 PM, Helmut Kuper wrote: > Hello, > > today I found that filtering events for a Job-UUID returned by a bgapi > command doesn't work well. > > What I do is this: > > $con->events("plain", "CHANNEL_HANGUP"); > $con->events("plain", "CHANNEL_ANSWER"); > $con->events("plain", "BACKGROUND_JOB"); > $con->filter("Unique-ID", $a_uuid); > $con->filter("Caller-Unique-ID", $a_uuid); > > $jobid=$con->bgapi("blahblah", ...)->getHeader("Job-UUID"); > $con->filter("Job-UUID", $jobid); > > Then waiting for events. But no BACKGROUND_JOB event is received. > Perhaps the BACKGROUND_JOB event is fired before the filter can be applied. > > > When I execute "$con->filter("Event-Name", "BACKGROUND_JOB");" before > "$jobid=..." then I'm able to receive it ... and all other but unwanted > BACKGROUND_JOB events as well. > > > Any trick to avoid this? > > regards > Helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Jun 29 00:21:13 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 29 Jun 2010 03:21:13 -0400 Subject: [Freeswitch-users] ESL install error In-Reply-To: <4C22AA59.8030007@nxivm.com> References: <4C22AA59.8030007@nxivm.com> Message-ID: <6593837863496108599@unknownmsgid> this is a bug in the php headers. please report this to php or your distro package maintainers for a proper fix. On Jun 23, 2010, at 8:44 PM, Steve O wrote: > Hello all, > > I'm trying to add phpmod-install into my Freeswitch installation on 2 > different servers (was failing on the first server, so trying another). > Here's a snippet of the error I'm encountering on both servers: > > g++ -I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb > -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable > -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM > -I/usr/include/php5/Zend -I/usr/include/php5/ext > -I/usr/include/php5/ext/date/lib -Wno-unused-label -Wno-unused-function > -c esl_wrap.cpp -o esl_wrap.o > cc1plus: warnings being treated as errors > esl_wrap.cpp: In function ?void _wrap_ESLevent_event_set(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1047: error: format not a string literal and no format > arguments > esl_wrap.cpp: In function ?void _wrap_ESLevent_event_get(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1073: error: format not a string literal and no format > arguments > esl_wrap.cpp: In function ?void > _wrap_ESLevent_serialized_string_set(int, zval*, zval**, zval*, int)?: > . > . > . > make[1]: *** [esl_wrap.o] Error 1 > make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/php' > make: *** [phpmod] Error 2 > > I searched the list for posts related to the [esl_wrap.o] Error 1, > followed all the advice that helped others, but am still having the > problem. A couple things about my situation I'm considering that might > be unique, and the problem (pretty much guessing at this point), is the > OS and architectures I'm running. One server's (the 1st server, > referenced above) running FreeBSD 8.0 on a 32 bit processor, and the > other server's running Ubuntu 10.04 on a 64 bit processor. > > Thoughts? What other information can I provide to help troubleshoot? > > Thanks, > > Steve > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 helmut.kuper at ewetel.de Tue Jun 29 02:29:26 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 29 Jun 2010 11:29:26 +0200 Subject: [Freeswitch-users] ESL: Question about BACKGROUND_JOB events In-Reply-To: <4274374276864666200@unknownmsgid> References: <4C22301C.3080405@ewetel.de> <4274374276864666200@unknownmsgid> Message-ID: <4C29BCF6.90701@ewetel.de> Hi Michael, erm, pardon? On 29.06.2010 09:13, Michael Jerris wrote: > origination_uuid > From uzairsh at yahoo.com Tue Jun 29 02:41:41 2010 From: uzairsh at yahoo.com (Syed Hussain) Date: Tue, 29 Jun 2010 02:41:41 -0700 (PDT) Subject: [Freeswitch-users] IP SIP Trunk errors/issues In-Reply-To: Message-ID: <134961.98667.qm@web30505.mail.mud.yahoo.com> I have tried the options below with no luck. David? will tcpdump output suffice ? please advise thanks S Re: [Freeswitch-users] IP SIP Trunk errors/issues Monday, June 28, 2010 8:12 AM From: "Zuhair Raza" To: freeswitch-users at lists.freeswitch.orghi try this one ? ???? On Mon, Jun 28, 2010 at 4:16 PM, Syed Hussain wrote: Hi , I have subscribed to sonovoip a sip trunk provider who uses IP based authentication for SIP services , I am trying to make outbound international calls which are being rejected by the provider for some reason, I am not able to figure out if there is something wrong with the configuration. I am listing out my configuration. Any help ?would be greatly appreciated . I have also tried #wiki.freeswitch.org/wiki/Provider_Configuration:_SonoVoIP configuration from freeswitch wiki gives errors . The voip configuration works perfectly with asterisk 1.6 My configuration is follows Under /usr/local/freeswitch/conf/sip_profiles/external cat sonovoip.xml ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? My dialplan , under /usr/local/freeswitch/conf/dialplan/public.xml ? ? ? ? ? ? ? ? ? ? ? ? I did change the vars.xml ?ports ? ? ? ? ? ? ? ? ? ? ? ? The debug #pastebin.com/4MFhSWm9 Please advise if any other information is required. Thanks S _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/e045ccd0/attachment.html From steveu at coppice.org Tue Jun 29 06:24:59 2010 From: steveu at coppice.org (Steve Underwood) Date: Tue, 29 Jun 2010 21:24:59 +0800 Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? In-Reply-To: <180EC6E8D9D94550A56D95A86BD3EDF2@mbnet.local> References: <328A03C01F9D4CE3973F4C86CB09FD7F@mbnet.local> <003e01cb16f2$09c7d920$1d578b60$@maly@molcs.org> <180EC6E8D9D94550A56D95A86BD3EDF2@mbnet.local> Message-ID: <4C29F42B.10403@coppice.org> On 06/29/2010 03:33 AM, Jeroen C. van Gelderen wrote: > Hi Mark, > > I'm curious to see if you'd had any luck with DAHDI built-in echo cancellers > such as MG2? (I figured since MG2 has been built into DAHDI for a while it > would be slightly more likely to work than OSLEC. In my case neither MG2 nor > OSLEC works.) > MG2 has very little chance of working. OSLEC is the only open source canceller which actually works. > In order to determine whether echo cancellation is in use with DAHDI you can > check the output of lsdahdi: > > [root at elastix ~]# lsdahdi > ### Span 1: XBUS-00/XPD-00 "Xorcom XPD #00/00: FXS" (MASTER) > 1 FXS FXOKS (In use) (SWEC: OSLEC) (EC: OSLEC) > 2 FXS FXOKS (In use) (SWEC: OSLEC) > > This shows FXS port 1 in off-hook with OSLEC in-use. > FXS port 2 has OSLEC configured but is on-hook. > > If I were to run FreeSwitch the channel would be silent in this > configuration. > Regards, Steve From david.ponzone at gmail.com Tue Jun 29 06:51:07 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Tue, 29 Jun 2010 15:51:07 +0200 Subject: [Freeswitch-users] IP SIP Trunk errors/issues In-Reply-To: <134961.98667.qm@web30505.mail.mud.yahoo.com> References: <134961.98667.qm@web30505.mail.mud.yahoo.com> Message-ID: <5300D72F-FC47-4FF4-A6D0-6317B479250C@gmail.com> sure tcpdump to a file (tcpdump -w save.cap etc...) and then put the file somewhere where we can access it David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 29/06/2010 ? 11:41, Syed Hussain a ?crit : > > I have tried the options below with no luck. > > David will tcpdump output suffice ? > > please advise > > thanks > > S > Re: [Freeswitch-users] IP SIP Trunk errors/issues > > Monday, June 28, 2010 8:12 AM > From: > "Zuhair Raza" > To: > freeswitch-users at lists.freeswitch.org > hi > try this one > > > > > > On Mon, Jun 28, 2010 at 4:16 PM, Syed Hussain > wrote: > Hi , > > I have subscribed to sonovoip a sip trunk provider who uses IP based > authentication for SIP services , > > I am trying to make outbound international calls which are being > rejected by the provider for some reason, I am not able to figure > out if there is something wrong with the configuration. > > I am listing out my configuration. Any help would be greatly > appreciated . > I have also tried #wiki.freeswitch.org/wiki/ > Provider_Configuration:_SonoVoIP configuration from freeswitch wiki > gives errors . > > The voip configuration works perfectly with asterisk 1.6 > > My configuration is follows > > Under > /usr/local/freeswitch/conf/sip_profiles/external > > cat sonovoip.xml > > > > > > > > > > > > > My dialplan , under /usr/local/freeswitch/conf/dialplan/public.xml > > > > > > > > I did change the vars.xml ports > > > > > > > > > > > > > > > The debug > > #pastebin.com/4MFhSWm9 > > > Please advise if any other information is required. > > Thanks > > S > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Regards, > Zuhair Raza > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/d5926f0a/attachment-0001.html From svetikvoip at gmail.com Tue Jun 29 08:10:45 2010 From: svetikvoip at gmail.com (Svetik) Date: Tue, 29 Jun 2010 11:10:45 -0400 Subject: [Freeswitch-users] Upgraded from 1.0.4 pre8 to the latest Git tree. Skype does not work anymore. Message-ID: <4C2A0CF5.4090606@gmail.com> Hi, On weekend I have upgraded to the latest Git tree from 1.0.4 pre8 which I was running for a long time, year may be. Everything went smooth, except Skype does not work anymore. Basically I followed Download & Installation Guide (http://wiki.freeswitch.org/wiki/Installation_Guide) and Mod skypopen Skype Endpoint and Trunk (http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk). The only difference, I have added at the end of the modules.conf.xml file to load skypopen module. My impression is that mod_skypopen hangs during initialization and freeswitch or some of its parts are locked and do not respond properly. When I call any number, I am getting busy signal immediately after keying the number. Log file shows: 2010-06-28 18:58:40.334877 [CRIT] switch_core_session.c:1524 Throttle Error! 0 In my case freeswitch and skype are started automatically by the scripts that are basically recommended ones. I have tried to load mod_skypopen manually (I removed mod_skypopen from the modules.conf.xml file) and most of the time it hangs. 1. I start the computer, freeswitch and skype autostart, but freeswitch does load mod_skypopen (it is not in the modules.conf.xml) 2. I shutdown freeswitch and start it from the console and wait for its prompt. 3. I do load mod_skypopen from the prompt. 4. I do load mod_skypopen from the prompt. 5. At this time it never returns to the prompt and there is no ringtone, log file shows: 2010-06-28 18:16:45.296626 [NOTICE] switch_channel.c:776 New Channel sofia/internal/1000 at 192.168.0.120 [d64d747e-8302-11df-91f3-1d2722671cfc] Occasionally, it works, it happens only the first time I am loading the module after machine was rebooted, when I restart freeswitch after that and try to load module again it always hangs, so I have to restart computer. When it hangs, freeswitch can not be shutdown with freeswitch -stop, I have to actually kill freeswitch process in order to restart it. After restarting freeswitch and Skype as well it follows the same bad scenario and hangs. I have to restart computer, and after that sometimes it works. Scripts to manage skype and freeswitch work perfect with version 1.0.4 pre8, I can restart freeswitch and skype multiple time no problem. I still have 1.0.4 pre8 sitting around and if I switch to it, it works perfectly. I am running recommended version of Skype 2.0.72 Everything is run on the 1Ghz machine (512MB RAM) Any thoughts what is wrong with my configuration? Anything I can try to resolve this issue? Thank you, Igor Below are samples of logs for "bad" and "good" loads. ----------------------------------- bad load ------------------------------------------- freeswitch at pbx> load mod_sk2010-06-27 20:40:52.462365 [WARNING] sofia.c:3700 Ping succeeded voipms with code 200 - count -1/1/1, state UP ypopen 2010-06-27 20:41:01.721128 [DEBUG] mod_skypopen.c:1223 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1223 ][none ][-1,-1,-1] globals.debug=8 2010-06-27 20:41:01.724336 [DEBUG] mod_skypopen.c:1230 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1230 ][none ][-1,-1,-1] globals.dialplan=XML 2010-06-27 20:41:01.724336 [DEBUG] mod_skypopen.c:1227 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1227 ][none ][-1,-1,-1] globals.context=default 2010-06-27 20:41:01.724336 [DEBUG] mod_skypopen.c:1233 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1233 ][none ][-1,-1,-1] globals.destination=5000 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1236 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1236 ][none ][-1,-1,-1] globals.skype_user=Boltik 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1239 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1239 ][none ][-1,-1,-1] globals.report_incoming_chatmessages=true 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1242 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1242 ][none ][-1,-1,-1] globals.silent_mode=false 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1245 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1245 ][none ][-1,-1,-1] globals.write_silence_when_idle=true 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1341 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1341 ][none ][-1,-1,-1] interface_id=1 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1352 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1352 ][none ][-1,-1,-1] name=Boltik 2010-06-27 20:41:01.726369 [DEBUG] mod_skypopen.c:1358 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1358 ][none ][-1,-1,-1] Initialized XInitThreads! 2010-06-27 20:41:01.727392 [DEBUG] mod_skypopen.c:1381 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1381 ][Boltik ][-1, 0, 0] CONFIGURING interface_id=1 2010-06-27 20:41:01.727392 [DEBUG] mod_skypopen.c:1418 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1418 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].name=Boltik 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1421 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1421 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].context=default 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1424 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1424 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].dialplan=XML 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1427 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1427 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].destination=5000 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1430 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1430 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].X11_display=:101 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1433 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1433 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].skype_user=Boltik 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1436 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1436 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].report_incoming_chatmessages=1 2010-06-27 20:41:01.729539 [DEBUG] mod_skypopen.c:1439 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1439 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].silent_mode=0 2010-06-27 20:41:01.729539 [DEBUG] mod_skypopen.c:1442 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1442 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].write_silence_when_idle=1 2010-06-27 20:41:01.729539 [DEBUG] mod_skypopen.c:1445 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1445 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].setsockopt=0 2010-06-27 20:41:01.729539 [WARNING] mod_skypopen.c:1447 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][WARNINGA 1447 ][Boltik ][-1, 0, 0] STARTING interface_id=1 2010-06-27 20:41:01.730932 [DEBUG] skypopen_protocol.c:1594 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1594 ][Boltik ][-1, 0, 0] X Display ':101' opened 2010-06-27 20:41:01.730932 [DEBUG] skypopen_protocol.c:1536 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1536 ][none ][-1,-1,-1] Skype instance found with id #2097368 2010-06-27 20:41:01.833832 [DEBUG] mod_skypopen.c:1150 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1150 ][Boltik ][-1, 0, 0] In skypopen_signaling_thread_func: started, p=0xb5bbaab8 2010-06-27 20:41:01.834965 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||OK||| 2010-06-27 20:41:01.834965 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||PROTOCOL 7||| 2010-06-27 20:41:01.834965 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| 2010-06-27 20:41:01.834965 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||CURRENTUSERHANDLE Boltik||| 2010-06-27 20:41:01.834965 [DEBUG] skypopen_protocol.c:263 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 263 ][Boltik ][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: Boltik, skype_user: Boltik! 2010-06-27 20:41:01.835975 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||USERSTATUS ONLINE||| 2010-06-27 20:41:01.936026 [NOTICE] mod_skypopen.c:1472 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][NOTICA 1472 ][Boltik ][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2010-06-27 20:41:01.936026 [NOTICE] mod_skypopen.c:1481 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][NOTICA 1481 ][Boltik ][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=1, waiting 60 seconds for CURRENTUSERHANDLE==Boltik 2010-06-27 20:41:01.936026 [WARNING] mod_skypopen.c:1500 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][WARNINGA 1500 ][Boltik ][-1, 0, 0] Interface_id=1 is now STARTED, the Skype client to which we are connected gave us the correct CURRENTUSERHANDLE (Boltik) 2010-06-27 20:41:01.937122 [DEBUG] skypopen_protocol.c:1494 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1494 ][Boltik ][-1, 0, 0] SENDING: |||PROTOCOL 7|||| 2010-06-27 20:41:01.938438 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||PROTOCOL 7||| 2010-06-27 20:41:02.006171 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| 2010-06-27 20:41:02.008363 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||CURRENTUSERHANDLE Boltik||| 2010-06-27 20:41:02.008363 [DEBUG] skypopen_protocol.c:263 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 263 ][Boltik ][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: Boltik, skype_user: Boltik! 2010-06-27 20:41:02.009489 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||USERSTATUS ONLINE||| ---------------------------------------------------------------------------------------- -------------------------------- good load --------------------------------------- freeswitch at pbx>load mod_skypopen 2010-06-27 22:48:30.592753 [DEBUG] mod_skypopen.c:1223 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1223 ][none ][-1,-1,-1] globals.debug=8 2010-06-27 22:48:30.597741 [DEBUG] mod_skypopen.c:1230 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1230 ][none ][-1,-1,-1] globals.dialplan=XML 2010-06-27 22:48:30.597741 [DEBUG] mod_skypopen.c:1227 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1227 ][none ][-1,-1,-1] globals.context=default 2010-06-27 22:48:30.597741 [DEBUG] mod_skypopen.c:1233 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1233 ][none ][-1,-1,-1] globals.destination=5000 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1236 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1236 ][none ][-1,-1,-1] globals.skype_user=Boltik 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1239 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1239 ][none ][-1,-1,-1] globals.report_incoming_chatmessages=true 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1242 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1242 ][none ][-1,-1,-1] globals.silent_mode=false 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1245 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1245 ][none ][-1,-1,-1] globals.write_silence_when_idle=true 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1341 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1341 ][none ][-1,-1,-1] interface_id=1 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1352 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1352 ][none ][-1,-1,-1] name=Boltik 2010-06-27 22:48:30.599763 [DEBUG] mod_skypopen.c:1358 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1358 ][none ][-1,-1,-1] Initialized XInitThreads! 2010-06-27 22:48:30.600769 [DEBUG] mod_skypopen.c:1381 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1381 ][Boltik ][-1, 0, 0] CONFIGURING interface_id=1 2010-06-27 22:48:30.600769 [DEBUG] mod_skypopen.c:1418 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1418 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].name=Boltik 2010-06-27 22:48:30.601785 [DEBUG] mod_skypopen.c:1421 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1421 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].context=default 2010-06-27 22:48:30.601785 [DEBUG] mod_skypopen.c:1424 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1424 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].dialplan=XML 2010-06-27 22:48:30.609658 [DEBUG] mod_skypopen.c:1427 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1427 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].destination=5000 2010-06-27 22:48:30.609658 [DEBUG] mod_skypopen.c:1430 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1430 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].X11_display=:101 2010-06-27 22:48:30.609658 [DEBUG] mod_skypopen.c:1433 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1433 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].skype_user=Boltik 2010-06-27 22:48:30.609658 [DEBUG] mod_skypopen.c:1436 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1436 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].report_incoming_chatmessages=1 2010-06-27 22:48:30.610668 [DEBUG] mod_skypopen.c:1439 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1439 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].silent_mode=0 2010-06-27 22:48:30.610668 [DEBUG] mod_skypopen.c:1442 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1442 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].write_silence_when_idle=1 2010-06-27 22:48:30.610668 [DEBUG] mod_skypopen.c:1445 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1445 ][Boltik ][-1, 0, 0] interface_id=1 globals.SKYPOPEN_INTERFACES[interface_id].setsockopt=0 2010-06-27 22:48:30.610668 [WARNING] mod_skypopen.c:1447 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][WARNINGA 1447 ][Boltik ][-1, 0, 0] STARTING interface_id=1 2010-06-27 22:48:30.612003 [DEBUG] skypopen_protocol.c:1594 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1594 ][Boltik ][-1, 0, 0] X Display ':101' opened 2010-06-27 22:48:30.613019 [DEBUG] skypopen_protocol.c:1536 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1536 ][none ][-1,-1,-1] Skype instance found with id #2097368 2010-06-27 22:48:30.716767 [DEBUG] mod_skypopen.c:1150 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1150 ][Boltik ][-1, 0, 0] In skypopen_signaling_thread_func: started, p=0xb5bb7ab8 2010-06-27 22:48:30.716767 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||OK||| 2010-06-27 22:48:30.717802 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||PROTOCOL 7||| 2010-06-27 22:48:30.717802 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| 2010-06-27 22:48:30.717802 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||CURRENTUSERHANDLE Boltik||| 2010-06-27 22:48:30.717802 [DEBUG] skypopen_protocol.c:263 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 263 ][Boltik ][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: Boltik, skype_user: Boltik! 2010-06-27 22:48:30.717802 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||USERSTATUS NA||| 2010-06-27 22:48:30.817902 [NOTICE] mod_skypopen.c:1472 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][NOTICA 1472 ][Boltik ][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2010-06-27 22:48:30.817902 [NOTICE] mod_skypopen.c:1481 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][NOTICA 1481 ][Boltik ][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=1, waiting 60 seconds for CURRENTUSERHANDLE==Boltik 2010-06-27 22:48:30.817902 [WARNING] mod_skypopen.c:1500 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][WARNINGA 1500 ][Boltik ][-1, 0, 0] Interface_id=1 is now STARTED, the Skype client to which we are connected gave us the correct CURRENTUSERHANDLE (Boltik) 2010-06-27 22:48:30.817902 [DEBUG] skypopen_protocol.c:1494 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1494 ][Boltik ][-1, 0, 0] SENDING: |||PROTOCOL 7|||| 2010-06-27 22:48:30.817902 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||PROTOCOL 7||| 2010-06-27 22:48:30.829711 [DEBUG] skypopen_protocol.c:1494 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1494 ][Boltik ][-1, 0, 0] SENDING: |||SET AUTOAWAY OFF|||| 2010-06-27 22:48:30.829711 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||||| 2010-06-27 22:48:30.840990 [DEBUG] skypopen_protocol.c:1494 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1494 ][Boltik ][-1, 0, 0] SENDING: |||SET WINDOWSTATE HIDDEN|||| 2010-06-27 22:48:30.840990 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||WINDOWSTATE HIDDEN||| 2010-06-27 22:48:30.897291 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| 2010-06-27 22:48:30.901487 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||CURRENTUSERHANDLE Boltik||| 2010-06-27 22:48:30.901487 [DEBUG] skypopen_protocol.c:263 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 263 ][Boltik ][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: Boltik, skype_user: Boltik! 2010-06-27 22:48:30.902499 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||USERSTATUS NA||| 2010-06-27 22:48:30.903510 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||AUTOAWAY OFF||| 2010-06-27 22:48:30.905503 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||USER Boltik ONLINESTATUS ONLINE||| 2010-06-27 22:48:30.906524 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||USERSTATUS ONLINE||| 2010-06-27 22:48:30.906524 [DEBUG] skypopen_protocol.c:1494 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1494 ][Boltik ][-1, 0, 0] SENDING: |||SET USERSTATUS ONLINE|||| 2010-06-27 22:48:30.908668 [DEBUG] skypopen_protocol.c:176 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 176 ][Boltik ][-1, 0, 0] READING: |||USERSTATUS ONLINE||| +OK 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1544 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1544 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].interface_id=1 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1545 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1545 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].name=Boltik 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1546 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1546 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].context=default 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1547 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1547 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].dialplan=XML 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1548 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1548 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].destination=5000 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1549 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1549 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].X11_display=:101 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1550 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1550 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].skype_user=Boltik 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1552 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1552 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].report_incoming_chatmessages=1 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1553 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1553 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].silent_mode=0 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1554 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1554 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].write_silence_when_idle=1 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1555 rev git2svn-syncpoint-master-121-g4e82098[(nil)|37 ][DEBUG_SKYPE 1555 ][Boltik ][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].setsockopt=0 2010-06-27 22:48:30.916757 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_skypopen] 2010-06-27 22:48:30.916757 [NOTICE] switch_loadable_module.c:145 Adding Endpoint 'skypopen' 2010-06-27 22:48:30.916757 [NOTICE] switch_loadable_module.c:273 Adding API Function 'sk' 2010-06-27 22:48:30.916757 [NOTICE] switch_loadable_module.c:273 Adding API Function 'skypopen' 2010-06-27 22:48:30.916757 [NOTICE] switch_loadable_module.c:273 Adding API Function 'skypopen_chat' 2010-06-27 22:48:30.916757 [NOTICE] switch_loadable_module.c:378 Adding Chat interface 'skype' freeswitch at pbx> ---------------------------------------------------------------------------------------- From sameer2k3t at gmail.com Tue Jun 29 09:11:54 2010 From: sameer2k3t at gmail.com (Sameer Khan) Date: Tue, 29 Jun 2010 21:11:54 +0500 Subject: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen Message-ID: Hi Can mod_dingaling be used for multiple outgoing calls to google network like skypopen.? If yes please reply -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/f2e0fee9/attachment.html From deya787 at gmail.com Tue Jun 29 09:31:13 2010 From: deya787 at gmail.com (Deya M) Date: Tue, 29 Jun 2010 19:31:13 +0300 Subject: [Freeswitch-users] Freeswitch Leakage ? Message-ID: HI all, Not sure if it is a leakage! Running FS, the memory used increases in a steady rate. I ran vg, and I can see a warning in the file, about an invalid file descriptor -1 in syscall close. http://pastebin.freeswitch.org/13298 http://pastebin.freeswitch.org/13299 Appreciate your help, D:- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/1bc5fd2b/attachment-0001.html From brian at freeswitch.org Tue Jun 29 09:40:21 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Jun 2010 11:40:21 -0500 Subject: [Freeswitch-users] Freeswitch Leakage ? In-Reply-To: References: Message-ID: If you ran valgrind properly you would have seen a leak report. valgrind --tool=memcheck --log-file-exactly=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes freeswitch -vg Then see the report. I bet you won't find much if any leaking. The memory usage will go up because we use memory pools. /b On Jun 29, 2010, at 11:31 AM, Deya M wrote: > HI all, > > Not sure if it is a leakage! > > Running FS, the memory used increases in a steady rate. > > I ran vg, and I can see a warning in the file, about an invalid file descriptor -1 in syscall close. > > > http://pastebin.freeswitch.org/13298 > http://pastebin.freeswitch.org/13299 > > Appreciate your help, > > D:- From ravi.kuru at callture.com Tue Jun 29 09:46:15 2010 From: ravi.kuru at callture.com (Ravi Kuru) Date: Tue, 29 Jun 2010 12:46:15 -0400 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 In-Reply-To: References: Message-ID: Hi Steve, I put install the jdk 64 bit now, i don't get that error but PhoneTest.java does not working i compile PhoneTest.java without error and freeswitch.log i see that trying to PhoneTest but it did not work. do you know why it is not working? this is the freeswitch.log: Dialplan: sofia/internal/6472585272 at 64.34.222.221 Regex (PASS) [public_did] destination_number(4166289121) =~ /^4166289121$/ break=on-false Dialplan: sofia/internal/6472585272 at 64.34.222.221 Action java(/usr/local/freeswitch/scripts/PhoneTest) 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/6472585272 at 64.34.222.221) State Change CS_ROUTING -> CS_EXECUTE 2010-06-29 11:48:50.496128 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/6472585272 at 64.34.222.221 [BREAK] 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/6472585272 at 64.34.222.221) State ROUTING going to sleep 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/6472585272 at 64.34.222.221) Running State Change CS_EXECUTE 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/6472585272 at 64.34.222.221) State EXECUTE 2010-06-29 11:48:50.496128 [DEBUG] mod_sofia.c:226 sofia/internal/ 6472585272 at 64.34.222.221 SOFIA EXECUTE 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:157 sofia/internal/6472585272 at 64.34.222.221 Standard EXECUTE EXECUTE sofia/internal/6472585272 at 64.34.222.221 set(outside_call=true) 2010-06-29 11:48:50.496128 [DEBUG] mod_dptools.c:816 sofia/internal/ 6472585272 at 64.34.222.221 SET [outside_call]=[true] EXECUTE sofia/internal/6472585272 at 64.34.222.221java(/usr/local/freeswitch/scripts/PhoneTest) 2010-06-29 11:48:50.496128 [NOTICE] switch_core_state_machine.c:185 sofia/internal/6472585272 at 64.34.222.221 has executed the last dialplan instruction, hanging up. 2010-06-29 11:48:50.496128 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/6472585272 at 64.34.222.221 [CS_EXECUTE] [NORMAL_CLEARING] 2010-06-29 11:48:50.496128 [DEBUG] switch_channel.c:2102 Send signal sofia/internal/6472585272 at 64.34.222.221 [KILL] 2010-06-29 11:48:50.496128 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/6472585272 at 64.34.222.221 [BREAK] 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/6472585272 at 64.34.222.221) State EXECUTE going to sleep 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/6472585272 at 64.34.222.221) Running State Change CS_HANGUP 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/6472585272 at 64.34.222.221) State HANGUP 2010-06-29 11:48:50.496128 [DEBUG] mod_sofia.c:414 Channel sofia/internal/ 6472585272 at 64.34.222.221 hanging up, cause: NORMAL_CLEARING 2010-06-29 11:48:50.498209 [DEBUG] mod_sofia.c:476 Responding to INVITE with: 480 2010-06-29 11:48:50.498209 [DEBUG] switch_core_state_machine.c:46 sofia/internal/6472585272 at 64.34.222.221 Standard HANGUP, cause: NORMAL_CLEARING 2010-06-29 11:48:50.498209 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/6472585272 at 64.34.222.221) State HANGUP going to sleep 2010-06-29 11:48:50.498209 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/6472585272 at 64.34.222.221) State Change CS_HANGUP -> CS_REPORTING 2010-06-29 11:48:50.500137 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/6472585272 at 64.34.222.221 [BREAK] 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/6472585272 at 64.34.222.221) Running State Change CS_REPORTING 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/6472585272 at 64.34.222.221) State REPORTING 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:53 sofia/internal/6472585272 at 64.34.222.221 Standard REPORTING, cause: NORMAL_CLEARING 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/6472585272 at 64.34.222.221) State REPORTING going to sleep 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/6472585272 at 64.34.222.221) State Change CS_REPORTING -> CS_DESTROY 2010-06-29 11:48:50.500138 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/6472585272 at 64.34.222.221 [BREAK] 2010-06-29 11:48:50.500138 [DEBUG] switch_core_session.c:1164 Session 7 (sofia/internal/6472585272 at 64.34.222.221) Locked, Waiting on external entities 2010-06-29 11:48:50.500138 [NOTICE] switch_core_session.c:1182 Session 7 (sofia/internal/6472585272 at 64.34.222.221) Ended 2010-06-29 11:48:50.500138 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/6472585272 at 64.34.222.221 [CS_DESTROY] 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/6472585272 at 64.34.222.221) Running State Change CS_DESTROY 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/6472585272 at 64.34.222.221) State DESTROY 2010-06-29 11:48:50.500138 [DEBUG] mod_sofia.c:341 sofia/internal/ 6472585272 at 64.34.222.221 SOFIA DESTROY 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:60 sofia/internal/6472585272 at 64.34.222.221 Standard DESTROY 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/6472585272 at 64.34.222.221) State DESTROY going to sleep import org.freeswitch.*; import org.freeswitch.swig.*; public class PhoneTest implements FreeswitchScript, DTMFCallback, HangupHook { public PhoneTest() { } public String onDTMF(Object object, int i, String arg) { if (object instanceof String) freeswitch.console_log("notice", "DTMF: " + (String)object + " ARG: " + arg + "\n"); else freeswitch.console_log("notice", "WOW GOT AN EVENT: " + object.toString()); return "true"; } public void onHangup() { freeswitch.console_log("notice", "HANGUP!\n"); } public void run(String sessionUuid, String args) { freeswitch.console_log("notice", "UUID: " + sessionUuid + " ARGS: " + args + "\n"); JavaSession session = null; String FileName = "/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_try_again.wav"; try { session = new JavaSession(sessionUuid); session.setDTMFCallback(this, "TEST"); session.setHangupHook(this); System.out.println("++++++++++++++++++++++++++++++++"); freeswitch.console_log("notice", "++++++++++Before Answer: +++++++++++++"); session.answer(); freeswitch.console_log("notice", "++++++++++After Answer: +++++++++++++"); //session.streamFile(FileName, 0); session.streamFile(FileName); freeswitch.console_log("notice", "++++++++++After play file: +++++++++++++" + FileName); session.recordFile("ravi.wav", 12000, 3000); session.hangup(""); } finally { if (session != null) session.delete(); } } } - Ravi On Fri, Jun 25, 2010 at 10:36 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: FreeSwitch Configuration : Help in SIP provider & gateway > configuration (Rupa Schomaker) > 2. Re: try to run mod_java but loading issue (Steven Ayre) > 3. Re: event-lock in mod_erlang_event (Andrew Thompson) > 4. Re: mod_erlang_event problem (Andrew Thompson) > 5. Re: Question about max-members (Michael Collins) > 6. ringback answer vs preanswer (Stephen Cattaneo) > 7. Re: ringback answer vs preanswer (Michael Collins) > 8. Re: Erlang Examples (Anthony Minessale) > > > ---------- Forwarded message ---------- > From: Rupa Schomaker > To: freeswitch-users > Date: Fri, 25 Jun 2010 08:49:19 -0500 > Subject: Re: [Freeswitch-users] FreeSwitch Configuration : Help in SIP > provider & gateway configuration > I'm a little confused. You say that you will be running without SIP (cards > + PRI) but then ask about setting up SIP. Which configuration are you > trying to do? > > On Fri, Jun 25, 2010 at 2:12 AM, Chaitanya Bhatt // Viva < > chaitanya at vivainfomedia.com> wrote: > >> Hey >> >> I have installed FreeSwitch successfully but not getting how to use it. I >> want to use FreeSwitch in Inbound/Outbound IVRS application. >> We will be using FreeSwitch with Sangoma Card & PRIs. In configuration >> section i am not getting SIP provider & gateway configuration. >> I am newbie in FreeSwitch, Can you please guide how to proceed with SIP >> provider & gateway configuration ? >> >> Incase of any further queries, Please feel free to mail me or contact me >> on the numbers provided below. >> >> Thanks & Regards, >> Chaitanya Bhatt >> Software Engineer. >> >> Viva Infomedia Pvt. Ltd. >> 242, Oshiwara Industrial Centre, >> New Link Road, Opp. Oshiwara Bus Depot, >> Goregaon West, Mumbai 400104. >> >> Direct: +91.22.40310356 >> Board: +91.22.40310310 >> Email : chaitanya at vivainfomedia.com >> >> Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India >> Awards 2009 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > > ---------- Forwarded message ---------- > From: Steven Ayre > To: freeswitch-users at lists.freeswitch.org > Date: Fri, 25 Jun 2010 15:13:41 +0100 > Subject: Re: [Freeswitch-users] try to run mod_java but loading issue > 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading > /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so > > Looks like it's a dependancy problem. Does that file exist, and are there > any other errors just before that line? > > Are you also on 32bit or 64bit? Looks like it's trying to load a 32bit > version (i386) which might be the problem if you're on 64bit FS. > > -Steve > > > On 24 June 2010 20:06, Ravi Kuru wrote: > >> Hi, >> >> I try to run PhoneTest on freeswitch and I followed the instruction but I >> got this error on freeswitch.log file when i start the freeswitch. >> >> 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading >> /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so >> 2010-06-24 15:01:47.096942 [CRIT] switch_loadable_module.c:882 Error >> Loading module /usr/local/freeswitch/mod/mod_java.so >> **Module load routine returned an error** >> >> what was the issue? >> >> Ravi >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Andrew Thompson > To: freeswitch-users at lists.freeswitch.org > Date: Fri, 25 Jun 2010 10:59:58 -0400 > Subject: Re: [Freeswitch-users] event-lock in mod_erlang_event > On Fri, Jun 25, 2010 at 10:41:37AM +0800, Seven Du wrote: > > Hi, > > > > As I can see, You cannot do this in inbound erlang > > > > sendmsg(FS, uuid, playback, "1.wav"); > > sendmsg(FS, uuid, playback, "2.wav"); > > sendmsg(FS, uuid, transfer, "xxxxx > > > > because it's async, and it will play 2.wav immediately. > > > > > > 1) Sure if I know the length of 1.wav I can > > > > sendmsg(FS, uuid, playback, 1.wav > > sleep(3000 > > sendmsg(FS, uuid, playback, 2.wav > > > > 2) Or I could wait the execute_complete event which will be a little > complicated > > > > > > According to http://wiki.freeswitch.org/wiki/Event_socket_outbound > > > > Is it possible to send a event-lock param to lock the message > temporarily? like > > > > sendmsg(FS, UUID, App, Args, [{"event-lock", "true"}]). > > > > Yes, event-lock works from erlang too (its implemented down in the core > I think). I use it to prevent exactly this problem. > > Feel free to update the wiki to note this is available. > > Andrew > > > > > ---------- Forwarded message ---------- > From: Andrew Thompson > To: freeswitch-users at lists.freeswitch.org > Date: Fri, 25 Jun 2010 11:06:22 -0400 > Subject: Re: [Freeswitch-users] mod_erlang_event problem > On Fri, Jun 25, 2010 at 09:32:44AM +0500, Timur Irmatov wrote: > > Hi, Andrew! > > That's the way it is done on my other server. And it works there fine. > > But, also, on that server I also use RPC mechanism the same way I done > > it on new one, and it works there fine too. It's just older freeswitch > > and older erlang there. Don't know if that matters, or may be RPC > > works there simply because I also use registered process and get_pid. > > So you use both mechanisms on a box with older freeswitch and erlang and > both work fine? Can you narrow it down by matching versions on one side > (freeswitch might be best) and see if the bug goes away/comes back? Its > possible the new erlang releases have changed something in ei, but its > more likely that I broke something in FreeSWITCH. What are the versions > of FS/erlang on the box that works? > > Andrew > > > > > ---------- Forwarded message ---------- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > Date: Fri, 25 Jun 2010 10:23:49 -0700 > Subject: Re: [Freeswitch-users] Question about max-members > For any conference created based in this profile it limits the total number > of conference participants. So if your conference profile has max-members=5 > then any conference you create based on that profile will allow only 5 > people to join the conference. > > -MC > > On Fri, Jun 25, 2010 at 1:41 AM, Sergey Scheglov wrote: > >> >> Hi All, >> >> max-members limits members in conference room or limits members using >> profile (i.e. default)? >> >> Regards >> Sergey Scheglov >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > ---------- Forwarded message ---------- > From: Stephen Cattaneo > To: freeswitch-users > Date: Fri, 25 Jun 2010 18:35:31 -0400 > Subject: [Freeswitch-users] ringback answer vs preanswer > i am trying to understand why when i do session.preanswer ringback works > when i bridge the call but when i do session.answer it does not. > > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print > > > > ---------- Forwarded message ---------- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > Date: Fri, 25 Jun 2010 17:45:47 -0700 > Subject: Re: [Freeswitch-users] ringback answer vs preanswer > Because ringback occurs in early media. You need transfer ringback after > the call is answered. See this page for more info: > http://wiki.freeswitch.org/wiki/Variable_transfer_ringback > > -MC > > On Fri, Jun 25, 2010 at 3:35 PM, Stephen Cattaneo wrote: > >> i am trying to understand why when i do session.preanswer ringback works >> when i bridge the call but when i do session.answer it does not. >> >> >> Thanks, >> Stephen C >> -All of my email addresses go to the same place >> -Save Paper, think before you print >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Date: Fri, 25 Jun 2010 21:35:32 -0500 > Subject: Re: [Freeswitch-users] Erlang Examples > Hey Seven, is JP still working with you guys? > We haven't heard from him in a year or more... =D > > > On Thu, Jun 24, 2010 at 8:35 PM, Seven Du wrote: > >> Welcome to the Erlang world. >> >> Erlang was initially designed to write telecom softwares and I found >> gen_fsm is easy to use and very clear to describe business logic. I'm >> the idapted person. I updated the post in the bottom to be more >> complete. >> >> Another vote to OpenACD because it is written by the author of >> mod_erlang_event. :) >> >> 2010/6/25 Anthony Minessale : >> > The guy who wrote mod_erlang_event and a developer of OpenACD is the >> same >> > guy already here helping him namely Andrew. >> > >> > >> > On Thu, Jun 24, 2010 at 11:27 AM, Jan Berger >> wrote: >> >> >> >> Hi, The OpenACD guys are writing the ACD in Erlang and integrating to >> FS, >> >> so >> >> you might find something there. >> >> >> >> --- >> >> >> >> I don't know the Ericsson Language that well myself, but having had a >> look >> >> at it I decided to stay away from this technology. >> >> >> >> What I am doing is writing IVR's in Java, C# or even C++ and I decided >> to >> >> use vxml/ccxml to bring IVR capability into the standard dev >> environment >> >> so >> >> I can deal with business in a proper language. >> >> >> >> Jan >> >> >> >> -----Original Message----- >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> David >> >> Swardstrom >> >> Sent: 24. juni 2010 17:52 >> >> To: freeswitch-users >> >> Subject: [Freeswitch-users] Erlang Examples >> >> >> >> I have been using JavaScript to handle a Conferencing application that >> >> started >> >> with the conf-ivr.js example program but is significantly more complex. >> >> This has been fun even though I had never used JavaScript before this >> >> year. >> >> >> >> However, there are things that seem to not be possible using >> JavaScript. >> >> I need to interact with several web based applications for several >> reasons >> >> and also need to provide some time based interactions with FreeSwitch >> >> and/or >> >> artifacts (Database entries, files of recorded conferences, etc). >> >> >> >> We (RemoteLink) have decided that the best solution for this support is >> >> to use an Erlang program and mod_erlang_event. So now I need to learn >> >> another language. >> >> >> >> But one thing that I do not find one the FreeSWITCH site is any Erlang >> >> examples. >> >> Are there some sample programs available such as one that would look >> for >> >> a certain type of event and print it out? >> >> >> >> I have found some semi-samples in the freeswitch-users archives but am >> >> somewhat >> >> ambivalent about using any of these without permission. >> >> >> >> Regards, >> >> David Swardstrom >> >> (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.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 >> > >> > >> > >> > -- >> > 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 >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Blog: http://www.dujinfang.com >> Proj: http://www.freeswitch.org.cn >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/1e0a486e/attachment-0001.html From roger_salloum at shaw.ca Tue Jun 29 10:05:21 2010 From: roger_salloum at shaw.ca (Roger Salloum) Date: Tue, 29 Jun 2010 10:05:21 -0700 Subject: [Freeswitch-users] Dialplan handling on call fails In-Reply-To: <5D032079-D1B3-4B5B-8DA7-059FB8549D53@gmail.com> References: <96D730D0-57A6-4751-9194-1285B4DCC8D2@shaw.ca> <795F82FA-51D3-4232-875A-84D3AEDCEF2F@gmail.com> <5D032079-D1B3-4B5B-8DA7-059FB8549D53@gmail.com> Message-ID: Hi David, I tried and it is function the same, it will still do the transfer/next bridge when a 480 occurs after ringing for 2 minutes. I set continue_on_fail=102. ----- Original Message ----- From: David Ponzone Date: Monday, June 28, 2010 3:44 pm Subject: Re: [Freeswitch-users] Dialplan handling on call fails To: freeswitch-users at lists.freeswitch.org > Perhaps by using continue_on_fail and putting a transfer just > after? > the bridge, so you can jump to another extension/dialplan ? > > David Ponzone? Direction Technique > email: david.ponzone at ipeva.fr > tel:????? 01 74 03 18 97 > gsm:?? 06 66 98 76 34 > > Service Client IPeva > tel:????? 0811 46 26 26 > www.ipeva.fr? -?? www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis? > ? l'intention exclusive de ses destinataires. Toute utilisation > ou? > diffusion non autoris?e est interdite. Tout message ?lectronique > est? > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > titre? > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > n'?tes? > pas destinataire de ce message, merci de le d?truire > imm?diatement et? > d'avertir l'exp?diteur. > > > > > Le 28/06/2010 ? 23:44, Roger Salloum a ?crit : > > > Hi David, > > > > Thanks for clearing that up. I tried when the call establishes > and? > > it would hang up, which is controlled by hangup_after_bridge. > Do you? > > have any suggestions for how i would get the failover > without? > > requiring all the bridges to be within the same condition? So > that i? > > can get failover to a less best match? > > > > ----- Original Message ----- > > From: David Ponzone > > Date: Monday, June 28, 2010 12:36 pm > > Subject: Re: [Freeswitch-users] Dialplan handling on call fails > > To: freeswitch-users at lists.freeswitch.org > > > > > Roger, > > > > > > you're right, but I think there is a mix up here. > > > In your dialplan, all your bridge actions will be executed > > > because of > > > the conditions they are included in, not because you use > > > continue_on_fail. > > > > > > The common use of continue_on_fail looks like: > > > > > > > > > > > > > > > > > > > > > > > > David Ponzone? Direction Technique > > > email: david.ponzone at ipeva.fr > > > tel:????? 01 74 03 18 97 > > > gsm:?? 06 66 98 76 34 > > > > > > Service Client IPeva > > > tel:????? 0811 46 26 26 > > > www.ipeva.fr? -?? www.ipeva-studio.com > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > ?tablis > > > ? l'intention exclusive de ses destinataires. Toute utilisation > > > ou > > > diffusion non autoris?e est interdite. Tout message ?lectronique > > > est > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > > titre > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > > n'?tes > > > pas destinataire de ce message, merci de le d?truire > > > imm?diatement et > > > d'avertir l'exp?diteur. > > > > > > > > > > > > > > > David Ponzone? Direction Technique > > > email: david.ponzone at ipeva.fr > > > tel:????? 01 74 03 18 97 > > > gsm:?? 06 66 98 76 34 > > > > > > Service Client IPeva > > > tel:????? 0811 46 26 26 > > > www.ipeva.fr? -?? www.ipeva-studio.com > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > ?tablis > > > ? l'intention exclusive de ses destinataires. Toute utilisation > > > ou > > > diffusion non autoris?e est interdite. Tout message ?lectronique > > > est > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > > titre > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > > n'?tes > > > pas destinataire de ce message, merci de le d?truire > > > imm?diatement et > > > d'avertir l'exp?diteur. > > > > > > > > > > > > > > > Le 28/06/2010 ? 21:06, Roger Salloum a ?crit : > > > > > > > >From my understanding the break is used to handle what > > > happens on > > > > the evaluation of the condition. I have break=never as I want > > > it to > > > > try all rules for the extension even if the one above returns > > > true > > > > or false. > > > > > > > > ----- Original Message ----- > > > > From: David Ponzone > > > > Date: Monday, June 28, 2010 6:36 am > > > > Subject: Re: [Freeswitch-users] Dialplan handling on call fails > > > > To: freeswitch-users at lists.freeswitch.org > > > > > > > > > If you use continue_on_fail, I don't think you need to set > > > > > break="never". > > > > > > > > > > David Ponzone? Direction Technique > > > > > email: david.ponzone at ipeva.fr > > > > > tel:????? 01 74 03 18 97 > > > > > gsm:?? 06 66 98 76 34 > > > > > > > > > > Service Client IPeva > > > > > tel:????? 0811 46 26 26 > > > > > www.ipeva.fr? -?? www.ipeva-studio.com > > > > > > > > > > Ce message et toutes les pi?ces jointes sont > confidentiels et > > > > > ?tablis > > > > > ? l'intention exclusive de ses destinataires. Toute > utilisation> > > > ou > > > > > diffusion non autoris?e est interdite. Tout message > ?lectronique> > > > est > > > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? > > > > > au > > > > > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. > > > > > Si > > > > > vous n'?tes pas destinataire de ce message, merci de le > > > > > d?truire > > > > > imm?diatement et d'avertir l'exp?diteur. > > > > > > > > > > > > > > > > > > > > > > > > > Le 26/06/2010 ? 16:53, Roger Salloum a ?crit : > > > > > > > > > > > Hi, > > > > > > > > > > > > I'm trying to setup a dialplan such that if one particular > > > > > route > > > > > > fails it will try another.? However, I do not > want it to > > > > > try another > > > > > > route once it had recieved a 180/183 in response from a > > > > > gateway. I > > > > > > have not been able to determine how to accomplish this. > > > > > > > > > > > > For Example: > > > > > > > > > > > >?? > > > > expression="^(1000123.*)$" > > > > > > break="never"> > > > > > >?????? > > > > application="set" data="hangup_after_bridge=true"/> > > > > > >? > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > > >?????? > > > > application="bridge" data="sofia/gateway/carrier1/$1"/> > > > > > >?? > > > > > > > > > > > >?? > > > > expression="^(1000.*)$" > > > > > > break="never"> > > > > > >?????? > > > > application="set" > > > > > > data="hangup_after_bridge=RECOVER_ON_TIMER_EXPIRE"/> > > > > > >?????? > > > > application="bridge" data="sofia/gateway/carrier2/$1"/> > > > > > >?? > > > > > > > > > > > >?? > > > > expression="^(1.*)$" > > > > > > break="never"> > > > > > >?????? > > > > application="set" data="hangup_after_bridge=true"/> > > > > > > > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > > >???? > > > > data="sofia/gateway/carrier3/$1"/>>?? > > > > > > > > > > > > So when 10001234567 is dialled i will match all 3. I'd > > > like to > > > > > be > > > > > > able to try 1, if failed, try 2, if failed try 3. All > > > calls go > > > > > out > > > > > > via an outbound proxy. > > > > > > > > > > > > Using the above examples if the gateway does not > respond in > > > > > time, > > > > > > the proxy generates a 408 REQUEST TIMEOUT error > message. It > > > > > will > > > > > > then fail out and try the next route. However, when the > > > > > gateway > > > > > > responds with a 180/183 but there is no answer after 2 > minutes> > > > the > > > > > > proxy, will generate a 480 NO ANSWER (also tried a 408 > > > > > REQUEST > > > > > > TIMEOUT ). When Freeswitch receives this message it > fails, and > > > > > then > > > > > > attempts the third failure route. How do i prevent the > > > > > dialplan from > > > > > > continuing once it has received a 180/183 when no one > answers> > > > the > > > > > > phone? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > _______________________________________________ > > > > > > FreeSWITCH-users mailing list > > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.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 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/6a0b9da0/attachment-0001.html From msc at freeswitch.org Tue Jun 29 10:15:06 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Jun 2010 10:15:06 -0700 Subject: [Freeswitch-users] SIP header on only one fork of a bridge In-Reply-To: <16BEF2FB-23BC-48F5-8C06-48E1BF66DDAB@freeswitch.org> References: <4C28EDB8.6080609@communicatefreely.net> <5FACB1C4-4BB1-4233-9354-02369E449748@gmail.com> <16BEF2FB-23BC-48F5-8C06-48E1BF66DDAB@freeswitch.org> Message-ID: On Mon, Jun 28, 2010 at 12:42 PM, Brian West wrote: > You can use [] in front of each sofia/ channel. > > This syntax is explained at the top of the channel variables wiki page: http://wiki.freeswitch.org/wiki/Channel_Variables -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/bd34e797/attachment.html From freeswitch-users at digitaldan.com Tue Jun 29 10:15:48 2010 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 29 Jun 2010 11:15:48 -0600 (MDT) Subject: [Freeswitch-users] calls ending with MEDIA_TIMEOUT In-Reply-To: <15069228.2982.1277830296778.JavaMail.daniel@radio> Message-ID: <13495796.2997.1277831734222.JavaMail.daniel@radio> Hi guys, I have been running two freeswitch boxes (13754M) that answer calls from a cisco 5300 (both on the same network) and records them to disk with a small lua application. This has been working well for the past few months. I decided to upgrade one of them to trunk ( git-3fbd9e2 2010-06-11 11-08-51 -0500 ) and have run into a problem. Some calls will fail with a MEDIA_TIMEOUT after a few minutes, the time it takes to fail ranges from 4 minutes to 10 minutes, I don't have a full sip trace or pcap dump yet, I reverted back to the old freeswitch version (on the same hardware) and have not been able to reproduce it in a test environment yet ( I continue to try). Below are the relevant lines from the log files for one of the calls: 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2257 (sofia/external/nobody at 192.168.21.4) Callstate Change ACTIVE -> HANGUP 2010-06-23 07:42:19.033466 [NOTICE] mod_sofia.c:884 Hangup sofia/external/nobody at 192.168.21.4 [CS_EXECUTE] [MEDIA_TIMEOUT] 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2273 Send signal sofia/external/nobody at 192.168.21.4 [KILL] 2010-06-23 07:42:19.033466 [DEBUG] switch_core_session.c:1023 Send signal sofia/external/nobody at 192.168.21.4 [BREAK] 2010-06-23 07:42:19.033466 [DEBUG] switch_core_codec.c:146 sofia/external/nobody at 192.168.21.4 Restore previous codec PCMU:0. My configuration is bone stock, so the rtp timeout value is at 300, but I have some calls that have lasted only 4 minutes. One other piece of information is that on one of the recordings that was hung up after 4 minutes and 17 seconds the recorded file was only 24 seconds long (it stopped recording after the first 24 seconds) , so I'm assuming freeswitch did not think there were any rtp packets to record. Any ideas on where to start debugging this? I have setup a new freeswitch box connected to the same 5300 to reproduce, but have not been able to generate the call volume ( there where around 30 calls being recorded) yet, but I'm working on it. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/6c30f42a/attachment.html From msc at freeswitch.org Tue Jun 29 10:21:01 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Jun 2010 10:21:01 -0700 Subject: [Freeswitch-users] ESL: Question about BACKGROUND_JOB events In-Reply-To: <4C29BCF6.90701@ewetel.de> References: <4C22301C.3080405@ewetel.de> <4274374276864666200@unknownmsgid> <4C29BCF6.90701@ewetel.de> Message-ID: I think he means you can generate a uuid and use the {origination_uuid=xxx} prefix so that you know what uuid to look for before you even start dialing... -MC On Tue, Jun 29, 2010 at 2:29 AM, Helmut Kuper wrote: > Hi Michael, > > erm, pardon? > > > On 29.06.2010 09:13, Michael Jerris wrote: > > origination_uuid > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/31aa7017/attachment.html From msc at freeswitch.org Tue Jun 29 11:09:53 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Jun 2010 11:09:53 -0700 Subject: [Freeswitch-users] calls ending with MEDIA_TIMEOUT In-Reply-To: <13495796.2997.1277831734222.JavaMail.daniel@radio> References: <15069228.2982.1277830296778.JavaMail.daniel@radio> <13495796.2997.1277831734222.JavaMail.daniel@radio> Message-ID: Pastebin your dialplan and the lua script for starters. Also, is it the 5300 that is responding with the media timeout? -MC On Tue, Jun 29, 2010 at 10:15 AM, Dan wrote: > Hi guys, I have been running two freeswitch boxes (13754M) that answer > calls from a cisco 5300 (both on the same network) and records them to disk > with a small lua application. This has been working well for the past few > months. I decided to upgrade one of them to trunk ( git-3fbd9e2 2010-06-11 > 11-08-51 -0500 ) and have run into a problem. Some calls will fail with a > MEDIA_TIMEOUT after a few minutes, the time it takes to fail ranges from 4 > minutes to 10 minutes, I don't have a full sip trace or pcap dump yet, I > reverted back to the old freeswitch version (on the same hardware) and have > not been able to reproduce it in a test environment yet ( I continue to > try). Below are the relevant lines from the log files for one of the > calls: > > 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2257 (sofia/external/ > nobody at 192.168.21.4) Callstate Change ACTIVE -> HANGUP > 2010-06-23 07:42:19.033466 [NOTICE] mod_sofia.c:884 Hangup sofia/external/ > nobody at 192.168.21.4 [CS_EXECUTE] [MEDIA_TIMEOUT] > 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2273 Send signal > sofia/external/nobody at 192.168.21.4 [KILL] > 2010-06-23 07:42:19.033466 [DEBUG] switch_core_session.c:1023 Send signal > sofia/external/nobody at 192.168.21.4 [BREAK] > 2010-06-23 07:42:19.033466 [DEBUG] switch_core_codec.c:146 sofia/external/ > nobody at 192.168.21.4 Restore previous codec PCMU:0. > > My configuration is bone stock, so the rtp timeout value is at 300, but I > have some calls that have lasted only 4 minutes. One other piece of > information is that on one of the recordings that was hung up after 4 > minutes and 17 seconds the recorded file was only 24 seconds long (it > stopped recording after the first 24 seconds) , so I'm assuming freeswitch > did not think there were any rtp packets to record. > > Any ideas on where to start debugging this? I have setup a new freeswitch > box connected to the same 5300 to reproduce, but have not been able to > generate the call volume ( there where around 30 calls being recorded) yet, > but I'm working on it. > > Thanks! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/5820a3de/attachment.html From anthony.minessale at gmail.com Tue Jun 29 11:54:21 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Jun 2010 13:54:21 -0500 Subject: [Freeswitch-users] calls ending with MEDIA_TIMEOUT In-Reply-To: References: <15069228.2982.1277830296778.JavaMail.daniel@radio> <13495796.2997.1277831734222.JavaMail.daniel@radio> Message-ID: it's not 100% accurate in the media timeout. It would be too expensive to use actual timers, it uses the number of samples you should be getting from rtp and a number of loops where no media was received. Migrating from svn 13000 range to GIT is a big step and you may have to adjust to some new behaviors. media_timeout may not even have existed that long ago I don't recall. If you don't need media timeouts turn off the param or turn it up to longer. On Tue, Jun 29, 2010 at 1:09 PM, Michael Collins wrote: > Pastebin your dialplan and the lua script for starters. Also, is it the > 5300 that is responding with the media timeout? > -MC > > On Tue, Jun 29, 2010 at 10:15 AM, Dan wrote: > >> Hi guys, I have been running two freeswitch boxes (13754M) that answer >> calls from a cisco 5300 (both on the same network) and records them to disk >> with a small lua application. This has been working well for the past few >> months. I decided to upgrade one of them to trunk ( git-3fbd9e2 2010-06-11 >> 11-08-51 -0500 ) and have run into a problem. Some calls will fail with a >> MEDIA_TIMEOUT after a few minutes, the time it takes to fail ranges from 4 >> minutes to 10 minutes, I don't have a full sip trace or pcap dump yet, I >> reverted back to the old freeswitch version (on the same hardware) and have >> not been able to reproduce it in a test environment yet ( I continue to >> try). Below are the relevant lines from the log files for one of the >> calls: >> >> 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2257 (sofia/external/ >> nobody at 192.168.21.4) Callstate Change ACTIVE -> HANGUP >> 2010-06-23 07:42:19.033466 [NOTICE] mod_sofia.c:884 Hangup sofia/external/ >> nobody at 192.168.21.4 [CS_EXECUTE] [MEDIA_TIMEOUT] >> 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2273 Send signal >> sofia/external/nobody at 192.168.21.4 [KILL] >> 2010-06-23 07:42:19.033466 [DEBUG] switch_core_session.c:1023 Send signal >> sofia/external/nobody at 192.168.21.4 [BREAK] >> 2010-06-23 07:42:19.033466 [DEBUG] switch_core_codec.c:146 sofia/external/ >> nobody at 192.168.21.4 Restore previous codec PCMU:0. >> >> My configuration is bone stock, so the rtp timeout value is at 300, but I >> have some calls that have lasted only 4 minutes. One other piece of >> information is that on one of the recordings that was hung up after 4 >> minutes and 17 seconds the recorded file was only 24 seconds long (it >> stopped recording after the first 24 seconds) , so I'm assuming freeswitch >> did not think there were any rtp packets to record. >> >> Any ideas on where to start debugging this? I have setup a new freeswitch >> box connected to the same 5300 to reproduce, but have not been able to >> generate the call volume ( there where around 30 calls being recorded) yet, >> but I'm working on it. >> >> Thanks! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100629/5a7e4bf5/attachment-0001.html From mrene_lists at avgs.ca Tue Jun 29 12:26:14 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 29 Jun 2010 15:26:14 -0400 Subject: [Freeswitch-users] ESL: Question about BACKGROUND_JOB events In-Reply-To: References: <4C22301C.3080405@ewetel.de> <4274374276864666200@unknownmsgid> <4C29BCF6.90701@ewetel.de> Message-ID: <21D9FC45-0FA9-4454-97AD-010E150470FF@avgs.ca> Hrum.... [origination_uuid=xxx], it wont work in { } since its a per-leg param. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-06-29, at 1:21 PM, Michael Collins wrote: > I think he means you can generate a uuid and use the {origination_uuid=xxx} prefix so that you know what uuid to look for before you even start dialing... > > -MC > > On Tue, Jun 29, 2010 at 2:29 AM, Helmut Kuper wrote: > Hi Michael, > > erm, pardon? > > > On 29.06.2010 09:13, Michael Jerris wrote: > > origination_uuid > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100629/017709ea/attachment.html From david.ponzone at gmail.com Tue Jun 29 12:42:51 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Tue, 29 Jun 2010 21:42:51 +0200 Subject: [Freeswitch-users] Dialplan handling on call fails In-Reply-To: References: <96D730D0-57A6-4751-9194-1285B4DCC8D2@shaw.ca> <795F82FA-51D3-4232-875A-84D3AEDCEF2F@gmail.com> <5D032079-D1B3-4B5B-8DA7-059FB8549D53@gmail.com> Message-ID: Hm then I am out of my league :) Perhaps you could send your current dialplan so anyone more knowledgable than I am could help ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 29/06/2010 ? 19:05, Roger Salloum a ?crit : > Hi David, > > I tried and it is function the same, it will still do the transfer/ > next bridge when a 480 occurs after ringing for 2 minutes. I set > continue_on_fail=102. > > ----- Original Message ----- > From: David Ponzone > Date: Monday, June 28, 2010 3:44 pm > Subject: Re: [Freeswitch-users] Dialplan handling on call fails > To: freeswitch-users at lists.freeswitch.org > > > Perhaps by using continue_on_fail and putting a transfer just > > after > > the bridge, so you can jump to another extension/dialplan ? > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > ?tablis > > ? l'intention exclusive de ses destinataires. Toute utilisation > > ou > > diffusion non autoris?e est interdite. Tout message ?lectronique > > est > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > titre > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > n'?tes > > pas destinataire de ce message, merci de le d?truire > > imm?diatement et > > d'avertir l'exp?diteur. > > > > > > > > > > Le 28/06/2010 ? 23:44, Roger Salloum a ?crit : > > > > > Hi David, > > > > > > Thanks for clearing that up. I tried when the call establishes > > and > > > it would hang up, which is controlled by hangup_after_bridge. > > Do you > > > have any suggestions for how i would get the failover > > without > > > requiring all the bridges to be within the same condition? So > > that i > > > can get failover to a less best match? > > > > > > ----- Original Message ----- > > > From: David Ponzone > > > Date: Monday, June 28, 2010 12:36 pm > > > Subject: Re: [Freeswitch-users] Dialplan handling on call fails > > > To: freeswitch-users at lists.freeswitch.org > > > > > > > Roger, > > > > > > > > you're right, but I think there is a mix up here. > > > > In your dialplan, all your bridge actions will be executed > > > > because of > > > > the conditions they are included in, not because you use > > > > continue_on_fail. > > > > > > > > The common use of continue_on_fail looks like: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > David Ponzone Direction Technique > > > > email: david.ponzone at ipeva.fr > > > > tel: 01 74 03 18 97 > > > > gsm: 06 66 98 76 34 > > > > > > > > Service Client IPeva > > > > tel: 0811 46 26 26 > > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > > ?tablis > > > > ? l'intention exclusive de ses destinataires. Toute utilisation > > > > ou > > > > diffusion non autoris?e est interdite. Tout message ?lectronique > > > > est > > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > > > titre > > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > > > n'?tes > > > > pas destinataire de ce message, merci de le d?truire > > > > imm?diatement et > > > > d'avertir l'exp?diteur. > > > > > > > > > > > > > > > > > > > > David Ponzone Direction Technique > > > > email: david.ponzone at ipeva.fr > > > > tel: 01 74 03 18 97 > > > > gsm: 06 66 98 76 34 > > > > > > > > Service Client IPeva > > > > tel: 0811 46 26 26 > > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > > ?tablis > > > > ? l'intention exclusive de ses destinataires. Toute utilisation > > > > ou > > > > diffusion non autoris?e est interdite. Tout message ?lectronique > > > > est > > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > > > titre > > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > > > n'?tes > > > > pas destinataire de ce message, merci de le d?truire > > > > imm?diatement et > > > > d'avertir l'exp?diteur. > > > > > > > > > > > > > > > > > > > > Le 28/06/2010 ? 21:06, Roger Salloum a ?crit : > > > > > > > > > >From my understanding the break is used to handle what > > > > happens on > > > > > the evaluation of the condition. I have break=never as I want > > > > it to > > > > > try all rules for the extension even if the one above returns > > > > true > > > > > or false. > > > > > > > > > > ----- Original Message ----- > > > > > From: David Ponzone > > > > > Date: Monday, June 28, 2010 6:36 am > > > > > Subject: Re: [Freeswitch-users] Dialplan handling on call > fails > > > > > To: freeswitch-users at lists.freeswitch.org > > > > > > > > > > > If you use continue_on_fail, I don't think you need to set > > > > > > break="never". > > > > > > > > > > > > David Ponzone Direction Technique > > > > > > email: david.ponzone at ipeva.fr > > > > > > tel: 01 74 03 18 97 > > > > > > gsm: 06 66 98 76 34 > > > > > > > > > > > > Service Client IPeva > > > > > > tel: 0811 46 26 26 > > > > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > > > > > > > Ce message et toutes les pi?ces jointes sont > > confidentiels et > > > > > > ?tablis > > > > > > ? l'intention exclusive de ses destinataires. Toute > > utilisation> > > > ou > > > > > > diffusion non autoris?e est interdite. Tout message > > ?lectronique> > > > est > > > > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? > > > > > > au > > > > > > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. > > > > > > Si > > > > > > vous n'?tes pas destinataire de ce message, merci de le > > > > > > d?truire > > > > > > imm?diatement et d'avertir l'exp?diteur. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Le 26/06/2010 ? 16:53, Roger Salloum a ?crit : > > > > > > > > > > > > > Hi, > > > > > > > > > > > > > > I'm trying to setup a dialplan such that if one particular > > > > > > route > > > > > > > fails it will try another. However, I do not > > want it to > > > > > > try another > > > > > > > route once it had recieved a 180/183 in response from a > > > > > > gateway. I > > > > > > > have not been able to determine how to accomplish this. > > > > > > > > > > > > > > For Example: > > > > > > > > > > > > > > > > > > > expression="^(1000123.*)$" > > > > > > > break="never"> > > > > > > > > > > > > application="set" data="hangup_after_bridge=true"/> > > > > > > > > > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > > > > > > > > > application="bridge" data="sofia/gateway/carrier1/$1"/> > > > > > > > > > > > > > > > > > > > > > > > > > > expression="^(1000.*)$" > > > > > > > break="never"> > > > > > > > > > > > > application="set" > > > > > > > data="hangup_after_bridge=RECOVER_ON_TIMER_EXPIRE"/> > > > > > > > > > > > > application="bridge" data="sofia/gateway/carrier2/$1"/> > > > > > > > > > > > > > > > > > > > > > > > > > > expression="^(1.*)$" > > > > > > > break="never"> > > > > > > > > > > > > application="set" data="hangup_after_bridge=true"/> > > > > > > > > > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > > > > > > > > > data="sofia/gateway/carrier3/$1"/>> > > > > > > > > > > > > > > So when 10001234567 is dialled i will match all 3. I'd > > > > like to > > > > > > be > > > > > > > able to try 1, if failed, try 2, if failed try 3. All > > > > calls go > > > > > > out > > > > > > > via an outbound proxy. > > > > > > > > > > > > > > Using the above examples if the gateway does not > > respond in > > > > > > time, > > > > > > > the proxy generates a 408 REQUEST TIMEOUT error > > message. It > > > > > > will > > > > > > > then fail out and try the next route. However, when the > > > > > > gateway > > > > > > > responds with a 180/183 but there is no answer after 2 > > minutes> > > > the > > > > > > > proxy, will generate a 480 NO ANSWER (also tried a 408 > > > > > > REQUEST > > > > > > > TIMEOUT ). When Freeswitch receives this message it > > fails, and > > > > > > then > > > > > > > attempts the third failure route. How do i prevent the > > > > > > dialplan from > > > > > > > continuing once it has received a 180/183 when no one > > answers> > > > the > > > > > > > phone? > > > > > > > > > > > > > > Thanks, > > > > > > > > > > > > > > _______________________________________________ > > > > > > > FreeSWITCH-users mailing list > > > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > http://www.freeswitch.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 > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.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/20100629/f2d37537/attachment-0001.html From roger_salloum at shaw.ca Tue Jun 29 12:52:22 2010 From: roger_salloum at shaw.ca (Roger Salloum) Date: Tue, 29 Jun 2010 12:52:22 -0700 Subject: [Freeswitch-users] Dialplan handling on call fails In-Reply-To: References: <96D730D0-57A6-4751-9194-1285B4DCC8D2@shaw.ca> <795F82FA-51D3-4232-875A-84D3AEDCEF2F@gmail.com> <5D032079-D1B3-4B5B-8DA7-059FB8549D53@gmail.com> Message-ID: Well thanks for the help regardless David, it cleared up a few things and gave me a couple new ideas. My current dialplan is pretty much the one i sent. Its mostly a test case, where 1st call to gateway fails with no response, and the second one succeeds. I just wanted to handle the case where a no answer occurs so that it does not attempt to ring the next gateway in the list, which is currently occurring. Freeswitch even shows a response of no_user_answer, and not recover_on_timer_expiry in the logs and i had only set continue on fail to recover_on_timer_expiry, so i'm a bit stumped as to why its is continuing. ----- Original Message ----- From: David Ponzone Date: Tuesday, June 29, 2010 12:43 pm Subject: Re: [Freeswitch-users] Dialplan handling on call fails To: freeswitch-users at lists.freeswitch.org > Hm then I am out of my league :) > > Perhaps you could send your current dialplan so anyone > more? > knowledgable than I am could help ? > > David Ponzone? Direction Technique > email: david.ponzone at ipeva.fr > tel:????? 01 74 03 18 97 > gsm:?? 06 66 98 76 34 > > Service Client IPeva > tel:????? 0811 46 26 26 > www.ipeva.fr? -?? www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis? > ? l'intention exclusive de ses destinataires. Toute utilisation > ou? > diffusion non autoris?e est interdite. Tout message ?lectronique > est? > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > titre? > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > n'?tes? > pas destinataire de ce message, merci de le d?truire > imm?diatement et? > d'avertir l'exp?diteur. > > > > > Le 29/06/2010 ? 19:05, Roger Salloum a ?crit : > > > Hi David, > > > > I tried and it is function the same, it will still do the > transfer/ > > next bridge when a 480 occurs after ringing for 2 minutes. I > set? > > continue_on_fail=102. > > > > ----- Original Message ----- > > From: David Ponzone > > Date: Monday, June 28, 2010 3:44 pm > > Subject: Re: [Freeswitch-users] Dialplan handling on call fails > > To: freeswitch-users at lists.freeswitch.org > > > > > Perhaps by using continue_on_fail and putting a transfer just > > > after > > > the bridge, so you can jump to another extension/dialplan ? > > > > > > David Ponzone? Direction Technique > > > email: david.ponzone at ipeva.fr > > > tel:????? 01 74 03 18 97 > > > gsm:?? 06 66 98 76 34 > > > > > > Service Client IPeva > > > tel:????? 0811 46 26 26 > > > www.ipeva.fr? -?? www.ipeva-studio.com > > > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > > > ?tablis > > > ? l'intention exclusive de ses destinataires. Toute utilisation > > > ou > > > diffusion non autoris?e est interdite. Tout message ?lectronique > > > est > > > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au > > > titre > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > > n'?tes > > > pas destinataire de ce message, merci de le d?truire > > > imm?diatement et > > > d'avertir l'exp?diteur. > > > > > > > > > > > > > > > Le 28/06/2010 ? 23:44, Roger Salloum a ?crit : > > > > > > > Hi David, > > > > > > > > Thanks for clearing that up. I tried when the call establishes > > > and > > > > it would hang up, which is controlled by hangup_after_bridge. > > > Do you > > > > have any suggestions for how i would get the failover > > > without > > > > requiring all the bridges to be within the same condition? So > > > that i > > > > can get failover to a less best match? > > > > > > > > ----- Original Message ----- > > > > From: David Ponzone > > > > Date: Monday, June 28, 2010 12:36 pm > > > > Subject: Re: [Freeswitch-users] Dialplan handling on call fails > > > > To: freeswitch-users at lists.freeswitch.org > > > > > > > > > Roger, > > > > > > > > > > you're right, but I think there is a mix up here. > > > > > In your dialplan, all your bridge actions will be executed > > > > > because of > > > > > the conditions they are included in, not because you use > > > > > continue_on_fail. > > > > > > > > > > The common use of continue_on_fail looks like: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > David Ponzone? Direction Technique > > > > > email: david.ponzone at ipeva.fr > > > > > tel:????? 01 74 03 18 97 > > > > > gsm:?? 06 66 98 76 34 > > > > > > > > > > Service Client IPeva > > > > > tel:????? 0811 46 26 26 > > > > > www.ipeva.fr? -?? www.ipeva-studio.com > > > > > > > > > > Ce message et toutes les pi?ces jointes sont > confidentiels et > > > > > ?tablis > > > > > ? l'intention exclusive de ses destinataires. Toute > utilisation> > > > ou > > > > > diffusion non autoris?e est interdite. Tout message > ?lectronique> > > > est > > > > > susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au > > > > > titre > > > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > > > > n'?tes > > > > > pas destinataire de ce message, merci de le d?truire > > > > > imm?diatement et > > > > > d'avertir l'exp?diteur. > > > > > > > > > > > > > > > > > > > > > > > > > David Ponzone? Direction Technique > > > > > email: david.ponzone at ipeva.fr > > > > > tel:????? 01 74 03 18 97 > > > > > gsm:?? 06 66 98 76 34 > > > > > > > > > > Service Client IPeva > > > > > tel:????? 0811 46 26 26 > > > > > www.ipeva.fr? -?? www.ipeva-studio.com > > > > > > > > > > Ce message et toutes les pi?ces jointes sont > confidentiels et > > > > > ?tablis > > > > > ? l'intention exclusive de ses destinataires. Toute > utilisation> > > > ou > > > > > diffusion non autoris?e est interdite. Tout message > ?lectronique> > > > est > > > > > susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au > > > > > titre > > > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > > > > > n'?tes > > > > > pas destinataire de ce message, merci de le d?truire > > > > > imm?diatement et > > > > > d'avertir l'exp?diteur. > > > > > > > > > > > > > > > > > > > > > > > > > Le 28/06/2010 ? 21:06, Roger Salloum a ?crit : > > > > > > > > > > > >From my understanding the break is used to handle what > > > > > happens on > > > > > > the evaluation of the condition. I have break=never as > I want > > > > > it to > > > > > > try all rules for the extension even if the one above > returns> > > > true > > > > > > or false. > > > > > > > > > > > > ----- Original Message ----- > > > > > > From: David Ponzone > > > > > > Date: Monday, June 28, 2010 6:36 am > > > > > > Subject: Re: [Freeswitch-users] Dialplan handling on > call? > > fails > > > > > > To: freeswitch-users at lists.freeswitch.org > > > > > > > > > > > > > If you use continue_on_fail, I don't think you need > to set > > > > > > > break="never". > > > > > > > > > > > > > > David Ponzone? Direction Technique > > > > > > > email: david.ponzone at ipeva.fr > > > > > > > tel:????? 01 74 03 18 97 > > > > > > > gsm:?? 06 66 98 76 34 > > > > > > > > > > > > > > Service Client IPeva > > > > > > > tel:????? 0811 46 26 26 > > > > > > > www.ipeva.fr? -?? www.ipeva-studio.com > > > > > > > > > > > > > > Ce message et toutes les pi?ces jointes sont > > > confidentiels et > > > > > > > ?tablis > > > > > > > ? l'intention exclusive de ses destinataires. Toute > > > utilisation> > > > ou > > > > > > > diffusion non autoris?e est interdite. Tout message > > > ?lectronique> > > > est > > > > > > > susceptible d'alt?ration. IPeva d?cline toute > responsabilit?> > > > > > au > > > > > > > titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?.> > > > > > Si > > > > > > > vous n'?tes pas destinataire de ce message, merci de le > > > > > > > d?truire > > > > > > > imm?diatement et d'avertir l'exp?diteur. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Le 26/06/2010 ? 16:53, Roger Salloum a ?crit : > > > > > > > > > > > > > > > Hi, > > > > > > > > > > > > > > > > I'm trying to setup a dialplan such that if one > particular> > > > > > route > > > > > > > > fails it will try another.? However, I do not > > > want it to > > > > > > > try another > > > > > > > > route once it had recieved a 180/183 in response > from a > > > > > > > gateway. I > > > > > > > > have not been able to determine how to accomplish this. > > > > > > > > > > > > > > > > For Example: > > > > > > > > > > > > > > > >?? > > > > > > expression="^(1000123.*)$" > > > > > > > > break="never"> > > > > > > > >?????? > > > > > > application="set" data="hangup_after_bridge=true"/> > > > > > > > >? > > > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > > > > >?????? > > > > > > application="bridge" data="sofia/gateway/carrier1/$1"/> > > > > > > > >?? > > > > > > > > > > > > > > > >?? > > > > > > expression="^(1000.*)$" > > > > > > > > break="never"> > > > > > > > >?????? > > > > > > application="set" > > > > > > > > data="hangup_after_bridge=RECOVER_ON_TIMER_EXPIRE"/> > > > > > > > >?????? > > > > > > application="bridge" data="sofia/gateway/carrier2/$1"/> > > > > > > > >?? > > > > > > > > > > > > > > > >?? > > > > > > expression="^(1.*)$" > > > > > > > > break="never"> > > > > > > > >?????? > > > > > > application="set" data="hangup_after_bridge=true"/> > > > > > > > > > > > > > > > data="continue_on_fail=RECOVER_ON_TIMER_EXPIRE"/> > > > > > > > >???? > > > > > > data="sofia/gateway/carrier3/$1"/>> > > > > > > > > > > > > > > > > So when 10001234567 is dialled i will match all 3. I'd > > > > > like to > > > > > > > be > > > > > > > > able to try 1, if failed, try 2, if failed try 3. All > > > > > calls go > > > > > > > out > > > > > > > > via an outbound proxy. > > > > > > > > > > > > > > > > Using the above examples if the gateway does not > > > respond in > > > > > > > time, > > > > > > > > the proxy generates a 408 REQUEST TIMEOUT error > > > message. It > > > > > > > will > > > > > > > > then fail out and try the next route. However, > when the > > > > > > > gateway > > > > > > > > responds with a 180/183 but there is no answer > after 2 > > > minutes> > > > the > > > > > > > > proxy, will generate a 480 NO ANSWER (also tried a 408 > > > > > > > REQUEST > > > > > > > > TIMEOUT ). When Freeswitch receives this message it > > > fails, and > > > > > > > then > > > > > > > > attempts the third failure route. How do i prevent the > > > > > > > dialplan from > > > > > > > > continuing once it has received a 180/183 when no one > > > answers> > > > the > > > > > > > > phone? > > > > > > > > > > > > > > > > Thanks, > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > > FreeSWITCH-users mailing list > > > > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > > > > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > http://www.freeswitch.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 > > > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.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/20100629/b6e5a8cc/attachment-0001.html From msc at freeswitch.org Tue Jun 29 13:14:48 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Jun 2010 13:14:48 -0700 Subject: [Freeswitch-users] ESL: Question about BACKGROUND_JOB events In-Reply-To: <21D9FC45-0FA9-4454-97AD-010E150470FF@avgs.ca> References: <4C22301C.3080405@ewetel.de> <4274374276864666200@unknownmsgid> <4C29BCF6.90701@ewetel.de> <21D9FC45-0FA9-4454-97AD-010E150470FF@avgs.ca> Message-ID: On Tue, Jun 29, 2010 at 12:26 PM, Mathieu Rene wrote: > Hrum.... [origination_uuid=xxx], it wont work in { } since its a per-leg > param. > > mea culpa! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/8b771b74/attachment.html From jerry.richards at teotech.com Tue Jun 29 13:27:56 2010 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 29 Jun 2010 13:27:56 -0700 Subject: [Freeswitch-users] Instant Message Sent Only To First Registered Extension Message-ID: I have multiple endpoints per extension. I noticed if I send an Instant Message from a softphone, FS sends the IM only to the endpoint that is first in the registration list (as seen by pressing F9). Is there a way to send the IM to all registered endpoints of a given extension? Thanks, Jerry From brian at freeswitch.org Tue Jun 29 13:31:35 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Jun 2010 15:31:35 -0500 Subject: [Freeswitch-users] Instant Message Sent Only To First Registered Extension In-Reply-To: References: Message-ID: <357693A9-B031-44A6-AD0A-4CEA9FCEA49E@freeswitch.org> Not that I'm aware of... that would have to be extended to allow you to do this. /b On Jun 29, 2010, at 3:27 PM, Jerry Richards wrote: > > I have multiple endpoints per extension. I noticed if I send an Instant > Message from a softphone, FS sends the IM only to the endpoint that is first > in the registration list (as seen by pressing F9). Is there a way to send > the IM to all registered endpoints of a given extension? > > Thanks, > Jerry From deya787 at gmail.com Tue Jun 29 13:59:42 2010 From: deya787 at gmail.com (Deya M) Date: Tue, 29 Jun 2010 23:59:42 +0300 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 214 In-Reply-To: References: Message-ID: Hi Brian, File after shutting down fs is complete: Please Check: *http://pastebin.freeswitch.org/13305* D:- 2010/6/29 > 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: Freeswitch Leakage ? (Brian West) > 2. Re: FreeSWITCH-users Digest, Vol 48, Issue 194 (Ravi Kuru) > > > ---------- Forwarded message ---------- > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Date: Tue, 29 Jun 2010 11:40:21 -0500 > Subject: Re: [Freeswitch-users] Freeswitch Leakage ? > If you ran valgrind properly you would have seen a leak report. > > valgrind --tool=memcheck --log-file-exactly=vg.log --leak-check=full > --leak-resolution=high --show-reachable=yes freeswitch -vg > > Then see the report. I bet you won't find much if any leaking. The memory > usage will go up because we use memory pools. > > /b > > On Jun 29, 2010, at 11:31 AM, Deya M wrote: > > > HI all, > > > > Not sure if it is a leakage! > > > > Running FS, the memory used increases in a steady rate. > > > > I ran vg, and I can see a warning in the file, about an invalid file > descriptor -1 in syscall close. > > > > > > http://pastebin.freeswitch.org/13298 > > http://pastebin.freeswitch.org/13299 > > > > Appreciate your help, > > > > D:- > > > > > > ---------- Forwarded message ---------- > From: Ravi Kuru > To: freeswitch-users at lists.freeswitch.org > Date: Tue, 29 Jun 2010 12:46:15 -0400 > Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 > Hi Steve, > > I put install the jdk 64 bit now, i don't get that error but PhoneTest.java > does not working > > i compile PhoneTest.java without error and freeswitch.log i see that trying > to PhoneTest but it did not work. > do you know why it is not working? > > this is the freeswitch.log: > Dialplan: sofia/internal/6472585272 at 64.34.222.221 Regex (PASS) > [public_did] destination_number(4166289121) =~ /^4166289121$/ break=on-false > Dialplan: sofia/internal/6472585272 at 64.34.222.221 Action > java(/usr/local/freeswitch/scripts/PhoneTest) > 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:119 > (sofia/internal/6472585272 at 64.34.222.221) State Change CS_ROUTING -> > CS_EXECUTE > 2010-06-29 11:48:50.496128 [DEBUG] switch_core_session.c:1021 Send signal > sofia/internal/6472585272 at 64.34.222.221 [BREAK] > 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/6472585272 at 64.34.222.221) State ROUTING going to sleep > 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/6472585272 at 64.34.222.221) Running State Change CS_EXECUTE > 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/6472585272 at 64.34.222.221) State EXECUTE > 2010-06-29 11:48:50.496128 [DEBUG] mod_sofia.c:226 sofia/internal/ > 6472585272 at 64.34.222.221 SOFIA EXECUTE > 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:157 > sofia/internal/6472585272 at 64.34.222.221 Standard EXECUTE > EXECUTE sofia/internal/6472585272 at 64.34.222.221 set(outside_call=true) > 2010-06-29 11:48:50.496128 [DEBUG] mod_dptools.c:816 sofia/internal/ > 6472585272 at 64.34.222.221 SET [outside_call]=[true] > EXECUTE sofia/internal/6472585272 at 64.34.222.221java(/usr/local/freeswitch/scripts/PhoneTest) > 2010-06-29 11:48:50.496128 [NOTICE] switch_core_state_machine.c:185 > sofia/internal/6472585272 at 64.34.222.221 has executed the last dialplan > instruction, hanging up. > 2010-06-29 11:48:50.496128 [NOTICE] switch_core_state_machine.c:187 Hangup > sofia/internal/6472585272 at 64.34.222.221 [CS_EXECUTE] [NORMAL_CLEARING] > 2010-06-29 11:48:50.496128 [DEBUG] switch_channel.c:2102 Send signal > sofia/internal/6472585272 at 64.34.222.221 [KILL] > 2010-06-29 11:48:50.496128 [DEBUG] switch_core_session.c:1021 Send signal > sofia/internal/6472585272 at 64.34.222.221 [BREAK] > 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/6472585272 at 64.34.222.221) State EXECUTE going to sleep > 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/6472585272 at 64.34.222.221) Running State Change CS_HANGUP > 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:499 > (sofia/internal/6472585272 at 64.34.222.221) State HANGUP > 2010-06-29 11:48:50.496128 [DEBUG] mod_sofia.c:414 Channel sofia/internal/ > 6472585272 at 64.34.222.221 hanging up, cause: NORMAL_CLEARING > 2010-06-29 11:48:50.498209 [DEBUG] mod_sofia.c:476 Responding to INVITE > with: 480 > 2010-06-29 11:48:50.498209 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/6472585272 at 64.34.222.221 Standard HANGUP, cause: > NORMAL_CLEARING > 2010-06-29 11:48:50.498209 [DEBUG] switch_core_state_machine.c:499 > (sofia/internal/6472585272 at 64.34.222.221) State HANGUP going to sleep > 2010-06-29 11:48:50.498209 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/6472585272 at 64.34.222.221) State Change CS_HANGUP -> > CS_REPORTING > 2010-06-29 11:48:50.500137 [DEBUG] switch_core_session.c:1021 Send signal > sofia/internal/6472585272 at 64.34.222.221 [BREAK] > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/6472585272 at 64.34.222.221) Running State Change > CS_REPORTING > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:590 > (sofia/internal/6472585272 at 64.34.222.221) State REPORTING > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/6472585272 at 64.34.222.221 Standard REPORTING, cause: > NORMAL_CLEARING > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:590 > (sofia/internal/6472585272 at 64.34.222.221) State REPORTING going to sleep > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/6472585272 at 64.34.222.221) State Change CS_REPORTING -> > CS_DESTROY > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_session.c:1021 Send signal > sofia/internal/6472585272 at 64.34.222.221 [BREAK] > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_session.c:1164 Session 7 > (sofia/internal/6472585272 at 64.34.222.221) Locked, Waiting on external > entities > 2010-06-29 11:48:50.500138 [NOTICE] switch_core_session.c:1182 Session 7 > (sofia/internal/6472585272 at 64.34.222.221) Ended > 2010-06-29 11:48:50.500138 [NOTICE] switch_core_session.c:1184 Close > Channel sofia/internal/6472585272 at 64.34.222.221 [CS_DESTROY] > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:428 > (sofia/internal/6472585272 at 64.34.222.221) Running State Change CS_DESTROY > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:439 > (sofia/internal/6472585272 at 64.34.222.221) State DESTROY > 2010-06-29 11:48:50.500138 [DEBUG] mod_sofia.c:341 sofia/internal/ > 6472585272 at 64.34.222.221 SOFIA DESTROY > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/6472585272 at 64.34.222.221 Standard DESTROY > 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:439 > (sofia/internal/6472585272 at 64.34.222.221) State DESTROY going to sleep > > import org.freeswitch.*; > import org.freeswitch.swig.*; > > public class PhoneTest implements FreeswitchScript, DTMFCallback, > HangupHook > { > public PhoneTest() > { > } > > public String onDTMF(Object object, int i, String arg) > { > if (object instanceof String) > freeswitch.console_log("notice", "DTMF: " + (String)object + " > ARG: " + arg + "\n"); > else > freeswitch.console_log("notice", "WOW GOT AN EVENT: " + > object.toString()); > return "true"; > } > > public void onHangup() > { > freeswitch.console_log("notice", "HANGUP!\n"); > } > > public void run(String sessionUuid, String args) > { > freeswitch.console_log("notice", "UUID: " + sessionUuid + " ARGS: " > + args + "\n"); > JavaSession session = null; > String FileName = > "/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_try_again.wav"; > try > { > session = new JavaSession(sessionUuid); > session.setDTMFCallback(this, "TEST"); > session.setHangupHook(this); > System.out.println("++++++++++++++++++++++++++++++++"); > freeswitch.console_log("notice", "++++++++++Before Answer: > +++++++++++++"); > session.answer(); > freeswitch.console_log("notice", "++++++++++After Answer: > +++++++++++++"); > //session.streamFile(FileName, 0); > session.streamFile(FileName); > freeswitch.console_log("notice", "++++++++++After play > file: +++++++++++++" + FileName); > session.recordFile("ravi.wav", 12000, 3000); > session.hangup(""); > } > finally > { > if (session != null) > session.delete(); > } > } > } > > > - Ravi > On Fri, Jun 25, 2010 at 10:36 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: FreeSwitch Configuration : Help in SIP provider & gateway >> configuration (Rupa Schomaker) >> 2. Re: try to run mod_java but loading issue (Steven Ayre) >> 3. Re: event-lock in mod_erlang_event (Andrew Thompson) >> 4. Re: mod_erlang_event problem (Andrew Thompson) >> 5. Re: Question about max-members (Michael Collins) >> 6. ringback answer vs preanswer (Stephen Cattaneo) >> 7. Re: ringback answer vs preanswer (Michael Collins) >> 8. Re: Erlang Examples (Anthony Minessale) >> >> >> ---------- Forwarded message ---------- >> From: Rupa Schomaker >> To: freeswitch-users >> Date: Fri, 25 Jun 2010 08:49:19 -0500 >> Subject: Re: [Freeswitch-users] FreeSwitch Configuration : Help in SIP >> provider & gateway configuration >> I'm a little confused. You say that you will be running without SIP >> (cards + PRI) but then ask about setting up SIP. Which configuration are >> you trying to do? >> >> On Fri, Jun 25, 2010 at 2:12 AM, Chaitanya Bhatt // Viva < >> chaitanya at vivainfomedia.com> wrote: >> >>> Hey >>> >>> I have installed FreeSwitch successfully but not getting how to use it. I >>> want to use FreeSwitch in Inbound/Outbound IVRS application. >>> We will be using FreeSwitch with Sangoma Card & PRIs. In configuration >>> section i am not getting SIP provider & gateway configuration. >>> I am newbie in FreeSwitch, Can you please guide how to proceed with SIP >>> provider & gateway configuration ? >>> >>> Incase of any further queries, Please feel free to mail me or contact me >>> on the numbers provided below. >>> >>> Thanks & Regards, >>> Chaitanya Bhatt >>> Software Engineer. >>> >>> Viva Infomedia Pvt. Ltd. >>> 242, Oshiwara Industrial Centre, >>> New Link Road, Opp. Oshiwara Bus Depot, >>> Goregaon West, Mumbai 400104. >>> >>> Direct: +91.22.40310356 >>> Board: +91.22.40310310 >>> Email : chaitanya at vivainfomedia.com >>> >>> Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India >>> Awards 2009 >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> -Rupa >> >> >> ---------- Forwarded message ---------- >> From: Steven Ayre >> To: freeswitch-users at lists.freeswitch.org >> Date: Fri, 25 Jun 2010 15:13:41 +0100 >> Subject: Re: [Freeswitch-users] try to run mod_java but loading issue >> 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading >> /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so >> >> Looks like it's a dependancy problem. Does that file exist, and are there >> any other errors just before that line? >> >> Are you also on 32bit or 64bit? Looks like it's trying to load a 32bit >> version (i386) which might be the problem if you're on 64bit FS. >> >> -Steve >> >> >> On 24 June 2010 20:06, Ravi Kuru wrote: >> >>> Hi, >>> >>> I try to run PhoneTest on freeswitch and I followed the instruction but I >>> got this error on freeswitch.log file when i start the freeswitch. >>> >>> 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading >>> /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so >>> 2010-06-24 15:01:47.096942 [CRIT] switch_loadable_module.c:882 Error >>> Loading module /usr/local/freeswitch/mod/mod_java.so >>> **Module load routine returned an error** >>> >>> what was the issue? >>> >>> Ravi >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> ---------- Forwarded message ---------- >> From: Andrew Thompson >> To: freeswitch-users at lists.freeswitch.org >> Date: Fri, 25 Jun 2010 10:59:58 -0400 >> Subject: Re: [Freeswitch-users] event-lock in mod_erlang_event >> On Fri, Jun 25, 2010 at 10:41:37AM +0800, Seven Du wrote: >> > Hi, >> > >> > As I can see, You cannot do this in inbound erlang >> > >> > sendmsg(FS, uuid, playback, "1.wav"); >> > sendmsg(FS, uuid, playback, "2.wav"); >> > sendmsg(FS, uuid, transfer, "xxxxx >> > >> > because it's async, and it will play 2.wav immediately. >> > >> > >> > 1) Sure if I know the length of 1.wav I can >> > >> > sendmsg(FS, uuid, playback, 1.wav >> > sleep(3000 >> > sendmsg(FS, uuid, playback, 2.wav >> > >> > 2) Or I could wait the execute_complete event which will be a little >> complicated >> > >> > >> > According to http://wiki.freeswitch.org/wiki/Event_socket_outbound >> > >> > Is it possible to send a event-lock param to lock the message >> temporarily? like >> > >> > sendmsg(FS, UUID, App, Args, [{"event-lock", "true"}]). >> > >> >> Yes, event-lock works from erlang too (its implemented down in the core >> I think). I use it to prevent exactly this problem. >> >> Feel free to update the wiki to note this is available. >> >> Andrew >> >> >> >> >> ---------- Forwarded message ---------- >> From: Andrew Thompson >> To: freeswitch-users at lists.freeswitch.org >> Date: Fri, 25 Jun 2010 11:06:22 -0400 >> Subject: Re: [Freeswitch-users] mod_erlang_event problem >> On Fri, Jun 25, 2010 at 09:32:44AM +0500, Timur Irmatov wrote: >> > Hi, Andrew! >> > That's the way it is done on my other server. And it works there fine. >> > But, also, on that server I also use RPC mechanism the same way I done >> > it on new one, and it works there fine too. It's just older freeswitch >> > and older erlang there. Don't know if that matters, or may be RPC >> > works there simply because I also use registered process and get_pid. >> >> So you use both mechanisms on a box with older freeswitch and erlang and >> both work fine? Can you narrow it down by matching versions on one side >> (freeswitch might be best) and see if the bug goes away/comes back? Its >> possible the new erlang releases have changed something in ei, but its >> more likely that I broke something in FreeSWITCH. What are the versions >> of FS/erlang on the box that works? >> >> Andrew >> >> >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: freeswitch-users at lists.freeswitch.org >> Date: Fri, 25 Jun 2010 10:23:49 -0700 >> Subject: Re: [Freeswitch-users] Question about max-members >> For any conference created based in this profile it limits the total >> number of conference participants. So if your conference profile has >> max-members=5 then any conference you create based on that profile will >> allow only 5 people to join the conference. >> >> -MC >> >> On Fri, Jun 25, 2010 at 1:41 AM, Sergey Scheglov wrote: >> >>> >>> Hi All, >>> >>> max-members limits members in conference room or limits members using >>> profile (i.e. default)? >>> >>> Regards >>> Sergey Scheglov >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> ---------- Forwarded message ---------- >> From: Stephen Cattaneo >> To: freeswitch-users >> Date: Fri, 25 Jun 2010 18:35:31 -0400 >> Subject: [Freeswitch-users] ringback answer vs preanswer >> i am trying to understand why when i do session.preanswer ringback works >> when i bridge the call but when i do session.answer it does not. >> >> >> Thanks, >> Stephen C >> -All of my email addresses go to the same place >> -Save Paper, think before you print >> >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: freeswitch-users at lists.freeswitch.org >> Date: Fri, 25 Jun 2010 17:45:47 -0700 >> Subject: Re: [Freeswitch-users] ringback answer vs preanswer >> Because ringback occurs in early media. You need transfer ringback after >> the call is answered. See this page for more info: >> http://wiki.freeswitch.org/wiki/Variable_transfer_ringback >> >> -MC >> >> On Fri, Jun 25, 2010 at 3:35 PM, Stephen Cattaneo wrote: >> >>> i am trying to understand why when i do session.preanswer ringback works >>> when i bridge the call but when i do session.answer it does not. >>> >>> >>> Thanks, >>> Stephen C >>> -All of my email addresses go to the same place >>> -Save Paper, think before you print >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale >> To: freeswitch-users at lists.freeswitch.org >> Date: Fri, 25 Jun 2010 21:35:32 -0500 >> Subject: Re: [Freeswitch-users] Erlang Examples >> Hey Seven, is JP still working with you guys? >> We haven't heard from him in a year or more... =D >> >> >> On Thu, Jun 24, 2010 at 8:35 PM, Seven Du wrote: >> >>> Welcome to the Erlang world. >>> >>> Erlang was initially designed to write telecom softwares and I found >>> gen_fsm is easy to use and very clear to describe business logic. I'm >>> the idapted person. I updated the post in the bottom to be more >>> complete. >>> >>> Another vote to OpenACD because it is written by the author of >>> mod_erlang_event. :) >>> >>> 2010/6/25 Anthony Minessale : >>> > The guy who wrote mod_erlang_event and a developer of OpenACD is the >>> same >>> > guy already here helping him namely Andrew. >>> > >>> > >>> > On Thu, Jun 24, 2010 at 11:27 AM, Jan Berger >>> wrote: >>> >> >>> >> Hi, The OpenACD guys are writing the ACD in Erlang and integrating to >>> FS, >>> >> so >>> >> you might find something there. >>> >> >>> >> --- >>> >> >>> >> I don't know the Ericsson Language that well myself, but having had a >>> look >>> >> at it I decided to stay away from this technology. >>> >> >>> >> What I am doing is writing IVR's in Java, C# or even C++ and I decided >>> to >>> >> use vxml/ccxml to bring IVR capability into the standard dev >>> environment >>> >> so >>> >> I can deal with business in a proper language. >>> >> >>> >> Jan >>> >> >>> >> -----Original Message----- >>> >> From: freeswitch-users-bounces at lists.freeswitch.org >>> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>> David >>> >> Swardstrom >>> >> Sent: 24. juni 2010 17:52 >>> >> To: freeswitch-users >>> >> Subject: [Freeswitch-users] Erlang Examples >>> >> >>> >> I have been using JavaScript to handle a Conferencing application that >>> >> started >>> >> with the conf-ivr.js example program but is significantly more >>> complex. >>> >> This has been fun even though I had never used JavaScript before this >>> >> year. >>> >> >>> >> However, there are things that seem to not be possible using >>> JavaScript. >>> >> I need to interact with several web based applications for several >>> reasons >>> >> and also need to provide some time based interactions with FreeSwitch >>> >> and/or >>> >> artifacts (Database entries, files of recorded conferences, etc). >>> >> >>> >> We (RemoteLink) have decided that the best solution for this support >>> is >>> >> to use an Erlang program and mod_erlang_event. So now I need to learn >>> >> another language. >>> >> >>> >> But one thing that I do not find one the FreeSWITCH site is any Erlang >>> >> examples. >>> >> Are there some sample programs available such as one that would look >>> for >>> >> a certain type of event and print it out? >>> >> >>> >> I have found some semi-samples in the freeswitch-users archives but am >>> >> somewhat >>> >> ambivalent about using any of these without permission. >>> >> >>> >> Regards, >>> >> David Swardstrom >>> >> (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom >>> >> >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.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 >>> > >>> > >>> > >>> > -- >>> > 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 >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> >>> >>> -- >>> Blog: http://www.dujinfang.com >>> Proj: http://www.freeswitch.org.cn >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100629/ecdf73dd/attachment-0001.html From deya787 at gmail.com Tue Jun 29 14:53:12 2010 From: deya787 at gmail.com (Deya M) Date: Wed, 30 Jun 2010 00:53:12 +0300 Subject: [Freeswitch-users] FS Leakage - FreeSWITCH-users Digest, Vol 48, Issue 214 Message-ID: Brian, Another File: (Second Test) * * *http://pastebin.freeswitch.org/13307* I am running another third test. D:- On Tue, Jun 29, 2010 at 11:59 PM, Deya M wrote: > Hi Brian, > > File after shutting down fs is complete: > > Please Check: > > *http://pastebin.freeswitch.org/13305* > > > D:- > > 2010/6/29 > >> 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: Freeswitch Leakage ? (Brian West) >> 2. Re: FreeSWITCH-users Digest, Vol 48, Issue 194 (Ravi Kuru) >> >> >> ---------- Forwarded message ---------- >> From: Brian West >> To: freeswitch-users at lists.freeswitch.org >> Date: Tue, 29 Jun 2010 11:40:21 -0500 >> Subject: Re: [Freeswitch-users] Freeswitch Leakage ? >> If you ran valgrind properly you would have seen a leak report. >> >> valgrind --tool=memcheck --log-file-exactly=vg.log --leak-check=full >> --leak-resolution=high --show-reachable=yes freeswitch -vg >> >> Then see the report. I bet you won't find much if any leaking. The >> memory usage will go up because we use memory pools. >> >> /b >> >> On Jun 29, 2010, at 11:31 AM, Deya M wrote: >> >> > HI all, >> > >> > Not sure if it is a leakage! >> > >> > Running FS, the memory used increases in a steady rate. >> > >> > I ran vg, and I can see a warning in the file, about an invalid file >> descriptor -1 in syscall close. >> > >> > >> > http://pastebin.freeswitch.org/13298 >> > http://pastebin.freeswitch.org/13299 >> > >> > Appreciate your help, >> > >> > D:- >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Ravi Kuru >> To: freeswitch-users at lists.freeswitch.org >> Date: Tue, 29 Jun 2010 12:46:15 -0400 >> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 >> Hi Steve, >> >> I put install the jdk 64 bit now, i don't get that error but >> PhoneTest.java does not working >> >> i compile PhoneTest.java without error and freeswitch.log i see that >> trying to PhoneTest but it did not work. >> do you know why it is not working? >> >> this is the freeswitch.log: >> Dialplan: sofia/internal/6472585272 at 64.34.222.221 Regex (PASS) >> [public_did] destination_number(4166289121) =~ /^4166289121$/ break=on-false >> Dialplan: sofia/internal/6472585272 at 64.34.222.221 Action >> java(/usr/local/freeswitch/scripts/PhoneTest) >> 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:119 >> (sofia/internal/6472585272 at 64.34.222.221) State Change CS_ROUTING -> >> CS_EXECUTE >> 2010-06-29 11:48:50.496128 [DEBUG] switch_core_session.c:1021 Send signal >> sofia/internal/6472585272 at 64.34.222.221 [BREAK] >> 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/6472585272 at 64.34.222.221) State ROUTING going to sleep >> 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/6472585272 at 64.34.222.221) Running State Change CS_EXECUTE >> 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/6472585272 at 64.34.222.221) State EXECUTE >> 2010-06-29 11:48:50.496128 [DEBUG] mod_sofia.c:226 sofia/internal/ >> 6472585272 at 64.34.222.221 SOFIA EXECUTE >> 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:157 >> sofia/internal/6472585272 at 64.34.222.221 Standard EXECUTE >> EXECUTE sofia/internal/6472585272 at 64.34.222.221 set(outside_call=true) >> 2010-06-29 11:48:50.496128 [DEBUG] mod_dptools.c:816 sofia/internal/ >> 6472585272 at 64.34.222.221 SET [outside_call]=[true] >> EXECUTE sofia/internal/6472585272 at 64.34.222.221java(/usr/local/freeswitch/scripts/PhoneTest) >> 2010-06-29 11:48:50.496128 [NOTICE] switch_core_state_machine.c:185 >> sofia/internal/6472585272 at 64.34.222.221 has executed the last dialplan >> instruction, hanging up. >> 2010-06-29 11:48:50.496128 [NOTICE] switch_core_state_machine.c:187 Hangup >> sofia/internal/6472585272 at 64.34.222.221 [CS_EXECUTE] [NORMAL_CLEARING] >> 2010-06-29 11:48:50.496128 [DEBUG] switch_channel.c:2102 Send signal >> sofia/internal/6472585272 at 64.34.222.221 [KILL] >> 2010-06-29 11:48:50.496128 [DEBUG] switch_core_session.c:1021 Send signal >> sofia/internal/6472585272 at 64.34.222.221 [BREAK] >> 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/6472585272 at 64.34.222.221) State EXECUTE going to sleep >> 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/6472585272 at 64.34.222.221) Running State Change CS_HANGUP >> 2010-06-29 11:48:50.496128 [DEBUG] switch_core_state_machine.c:499 >> (sofia/internal/6472585272 at 64.34.222.221) State HANGUP >> 2010-06-29 11:48:50.496128 [DEBUG] mod_sofia.c:414 Channel sofia/internal/ >> 6472585272 at 64.34.222.221 hanging up, cause: NORMAL_CLEARING >> 2010-06-29 11:48:50.498209 [DEBUG] mod_sofia.c:476 Responding to INVITE >> with: 480 >> 2010-06-29 11:48:50.498209 [DEBUG] switch_core_state_machine.c:46 >> sofia/internal/6472585272 at 64.34.222.221 Standard HANGUP, cause: >> NORMAL_CLEARING >> 2010-06-29 11:48:50.498209 [DEBUG] switch_core_state_machine.c:499 >> (sofia/internal/6472585272 at 64.34.222.221) State HANGUP going to sleep >> 2010-06-29 11:48:50.498209 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/6472585272 at 64.34.222.221) State Change CS_HANGUP -> >> CS_REPORTING >> 2010-06-29 11:48:50.500137 [DEBUG] switch_core_session.c:1021 Send signal >> sofia/internal/6472585272 at 64.34.222.221 [BREAK] >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/6472585272 at 64.34.222.221) Running State Change >> CS_REPORTING >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:590 >> (sofia/internal/6472585272 at 64.34.222.221) State REPORTING >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:53 >> sofia/internal/6472585272 at 64.34.222.221 Standard REPORTING, cause: >> NORMAL_CLEARING >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:590 >> (sofia/internal/6472585272 at 64.34.222.221) State REPORTING going to sleep >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:327 >> (sofia/internal/6472585272 at 64.34.222.221) State Change CS_REPORTING -> >> CS_DESTROY >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_session.c:1021 Send signal >> sofia/internal/6472585272 at 64.34.222.221 [BREAK] >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_session.c:1164 Session 7 >> (sofia/internal/6472585272 at 64.34.222.221) Locked, Waiting on external >> entities >> 2010-06-29 11:48:50.500138 [NOTICE] switch_core_session.c:1182 Session 7 >> (sofia/internal/6472585272 at 64.34.222.221) Ended >> 2010-06-29 11:48:50.500138 [NOTICE] switch_core_session.c:1184 Close >> Channel sofia/internal/6472585272 at 64.34.222.221 [CS_DESTROY] >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:428 >> (sofia/internal/6472585272 at 64.34.222.221) Running State Change CS_DESTROY >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:439 >> (sofia/internal/6472585272 at 64.34.222.221) State DESTROY >> 2010-06-29 11:48:50.500138 [DEBUG] mod_sofia.c:341 sofia/internal/ >> 6472585272 at 64.34.222.221 SOFIA DESTROY >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:60 >> sofia/internal/6472585272 at 64.34.222.221 Standard DESTROY >> 2010-06-29 11:48:50.500138 [DEBUG] switch_core_state_machine.c:439 >> (sofia/internal/6472585272 at 64.34.222.221) State DESTROY going to sleep >> >> import org.freeswitch.*; >> import org.freeswitch.swig.*; >> >> public class PhoneTest implements FreeswitchScript, DTMFCallback, >> HangupHook >> { >> public PhoneTest() >> { >> } >> >> public String onDTMF(Object object, int i, String arg) >> { >> if (object instanceof String) >> freeswitch.console_log("notice", "DTMF: " + (String)object + " >> ARG: " + arg + "\n"); >> else >> freeswitch.console_log("notice", "WOW GOT AN EVENT: " + >> object.toString()); >> return "true"; >> } >> >> public void onHangup() >> { >> freeswitch.console_log("notice", "HANGUP!\n"); >> } >> >> public void run(String sessionUuid, String args) >> { >> freeswitch.console_log("notice", "UUID: " + sessionUuid + " ARGS: >> " + args + "\n"); >> JavaSession session = null; >> String FileName = >> "/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_try_again.wav"; >> try >> { >> session = new JavaSession(sessionUuid); >> session.setDTMFCallback(this, "TEST"); >> session.setHangupHook(this); >> System.out.println("++++++++++++++++++++++++++++++++"); >> freeswitch.console_log("notice", "++++++++++Before Answer: >> +++++++++++++"); >> session.answer(); >> freeswitch.console_log("notice", "++++++++++After Answer: >> +++++++++++++"); >> //session.streamFile(FileName, 0); >> session.streamFile(FileName); >> freeswitch.console_log("notice", "++++++++++After play >> file: +++++++++++++" + FileName); >> session.recordFile("ravi.wav", 12000, 3000); >> session.hangup(""); >> } >> finally >> { >> if (session != null) >> session.delete(); >> } >> } >> } >> >> >> - Ravi >> On Fri, Jun 25, 2010 at 10:36 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: FreeSwitch Configuration : Help in SIP provider & gateway >>> configuration (Rupa Schomaker) >>> 2. Re: try to run mod_java but loading issue (Steven Ayre) >>> 3. Re: event-lock in mod_erlang_event (Andrew Thompson) >>> 4. Re: mod_erlang_event problem (Andrew Thompson) >>> 5. Re: Question about max-members (Michael Collins) >>> 6. ringback answer vs preanswer (Stephen Cattaneo) >>> 7. Re: ringback answer vs preanswer (Michael Collins) >>> 8. Re: Erlang Examples (Anthony Minessale) >>> >>> >>> ---------- Forwarded message ---------- >>> From: Rupa Schomaker >>> To: freeswitch-users >>> Date: Fri, 25 Jun 2010 08:49:19 -0500 >>> Subject: Re: [Freeswitch-users] FreeSwitch Configuration : Help in SIP >>> provider & gateway configuration >>> I'm a little confused. You say that you will be running without SIP >>> (cards + PRI) but then ask about setting up SIP. Which configuration are >>> you trying to do? >>> >>> On Fri, Jun 25, 2010 at 2:12 AM, Chaitanya Bhatt // Viva < >>> chaitanya at vivainfomedia.com> wrote: >>> >>>> Hey >>>> >>>> I have installed FreeSwitch successfully but not getting how to use it. >>>> I want to use FreeSwitch in Inbound/Outbound IVRS application. >>>> We will be using FreeSwitch with Sangoma Card & PRIs. In configuration >>>> section i am not getting SIP provider & gateway configuration. >>>> I am newbie in FreeSwitch, Can you please guide how to proceed with SIP >>>> provider & gateway configuration ? >>>> >>>> Incase of any further queries, Please feel free to mail me or contact me >>>> on the numbers provided below. >>>> >>>> Thanks & Regards, >>>> Chaitanya Bhatt >>>> Software Engineer. >>>> >>>> Viva Infomedia Pvt. Ltd. >>>> 242, Oshiwara Industrial Centre, >>>> New Link Road, Opp. Oshiwara Bus Depot, >>>> Goregaon West, Mumbai 400104. >>>> >>>> Direct: +91.22.40310356 >>>> Board: +91.22.40310310 >>>> Email : chaitanya at vivainfomedia.com >>>> >>>> Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India >>>> Awards 2009 >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> -Rupa >>> >>> >>> ---------- Forwarded message ---------- >>> From: Steven Ayre >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Fri, 25 Jun 2010 15:13:41 +0100 >>> Subject: Re: [Freeswitch-users] try to run mod_java but loading issue >>> 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading >>> /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so >>> >>> Looks like it's a dependancy problem. Does that file exist, and are there >>> any other errors just before that line? >>> >>> Are you also on 32bit or 64bit? Looks like it's trying to load a 32bit >>> version (i386) which might be the problem if you're on 64bit FS. >>> >>> -Steve >>> >>> >>> On 24 June 2010 20:06, Ravi Kuru wrote: >>> >>>> Hi, >>>> >>>> I try to run PhoneTest on freeswitch and I followed the instruction but >>>> I got this error on freeswitch.log file when i start the freeswitch. >>>> >>>> 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading >>>> /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so >>>> 2010-06-24 15:01:47.096942 [CRIT] switch_loadable_module.c:882 Error >>>> Loading module /usr/local/freeswitch/mod/mod_java.so >>>> **Module load routine returned an error** >>>> >>>> what was the issue? >>>> >>>> Ravi >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Andrew Thompson >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Fri, 25 Jun 2010 10:59:58 -0400 >>> Subject: Re: [Freeswitch-users] event-lock in mod_erlang_event >>> On Fri, Jun 25, 2010 at 10:41:37AM +0800, Seven Du wrote: >>> > Hi, >>> > >>> > As I can see, You cannot do this in inbound erlang >>> > >>> > sendmsg(FS, uuid, playback, "1.wav"); >>> > sendmsg(FS, uuid, playback, "2.wav"); >>> > sendmsg(FS, uuid, transfer, "xxxxx >>> > >>> > because it's async, and it will play 2.wav immediately. >>> > >>> > >>> > 1) Sure if I know the length of 1.wav I can >>> > >>> > sendmsg(FS, uuid, playback, 1.wav >>> > sleep(3000 >>> > sendmsg(FS, uuid, playback, 2.wav >>> > >>> > 2) Or I could wait the execute_complete event which will be a little >>> complicated >>> > >>> > >>> > According to http://wiki.freeswitch.org/wiki/Event_socket_outbound >>> > >>> > Is it possible to send a event-lock param to lock the message >>> temporarily? like >>> > >>> > sendmsg(FS, UUID, App, Args, [{"event-lock", "true"}]). >>> > >>> >>> Yes, event-lock works from erlang too (its implemented down in the core >>> I think). I use it to prevent exactly this problem. >>> >>> Feel free to update the wiki to note this is available. >>> >>> Andrew >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Andrew Thompson >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Fri, 25 Jun 2010 11:06:22 -0400 >>> Subject: Re: [Freeswitch-users] mod_erlang_event problem >>> On Fri, Jun 25, 2010 at 09:32:44AM +0500, Timur Irmatov wrote: >>> > Hi, Andrew! >>> > That's the way it is done on my other server. And it works there fine. >>> > But, also, on that server I also use RPC mechanism the same way I done >>> > it on new one, and it works there fine too. It's just older freeswitch >>> > and older erlang there. Don't know if that matters, or may be RPC >>> > works there simply because I also use registered process and get_pid. >>> >>> So you use both mechanisms on a box with older freeswitch and erlang and >>> both work fine? Can you narrow it down by matching versions on one side >>> (freeswitch might be best) and see if the bug goes away/comes back? Its >>> possible the new erlang releases have changed something in ei, but its >>> more likely that I broke something in FreeSWITCH. What are the versions >>> of FS/erlang on the box that works? >>> >>> Andrew >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Michael Collins >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Fri, 25 Jun 2010 10:23:49 -0700 >>> Subject: Re: [Freeswitch-users] Question about max-members >>> For any conference created based in this profile it limits the total >>> number of conference participants. So if your conference profile has >>> max-members=5 then any conference you create based on that profile will >>> allow only 5 people to join the conference. >>> >>> -MC >>> >>> On Fri, Jun 25, 2010 at 1:41 AM, Sergey Scheglov wrote: >>> >>>> >>>> Hi All, >>>> >>>> max-members limits members in conference room or limits members using >>>> profile (i.e. default)? >>>> >>>> Regards >>>> Sergey Scheglov >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Stephen Cattaneo >>> To: freeswitch-users >>> Date: Fri, 25 Jun 2010 18:35:31 -0400 >>> Subject: [Freeswitch-users] ringback answer vs preanswer >>> i am trying to understand why when i do session.preanswer ringback works >>> when i bridge the call but when i do session.answer it does not. >>> >>> >>> Thanks, >>> Stephen C >>> -All of my email addresses go to the same place >>> -Save Paper, think before you print >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Michael Collins >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Fri, 25 Jun 2010 17:45:47 -0700 >>> Subject: Re: [Freeswitch-users] ringback answer vs preanswer >>> Because ringback occurs in early media. You need transfer ringback after >>> the call is answered. See this page for more info: >>> http://wiki.freeswitch.org/wiki/Variable_transfer_ringback >>> >>> -MC >>> >>> On Fri, Jun 25, 2010 at 3:35 PM, Stephen Cattaneo wrote: >>> >>>> i am trying to understand why when i do session.preanswer ringback works >>>> when i bridge the call but when i do session.answer it does not. >>>> >>>> >>>> Thanks, >>>> Stephen C >>>> -All of my email addresses go to the same place >>>> -Save Paper, think before you print >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Anthony Minessale >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Fri, 25 Jun 2010 21:35:32 -0500 >>> Subject: Re: [Freeswitch-users] Erlang Examples >>> Hey Seven, is JP still working with you guys? >>> We haven't heard from him in a year or more... =D >>> >>> >>> On Thu, Jun 24, 2010 at 8:35 PM, Seven Du wrote: >>> >>>> Welcome to the Erlang world. >>>> >>>> Erlang was initially designed to write telecom softwares and I found >>>> gen_fsm is easy to use and very clear to describe business logic. I'm >>>> the idapted person. I updated the post in the bottom to be more >>>> complete. >>>> >>>> Another vote to OpenACD because it is written by the author of >>>> mod_erlang_event. :) >>>> >>>> 2010/6/25 Anthony Minessale : >>>> > The guy who wrote mod_erlang_event and a developer of OpenACD is the >>>> same >>>> > guy already here helping him namely Andrew. >>>> > >>>> > >>>> > On Thu, Jun 24, 2010 at 11:27 AM, Jan Berger >>>> wrote: >>>> >> >>>> >> Hi, The OpenACD guys are writing the ACD in Erlang and integrating to >>>> FS, >>>> >> so >>>> >> you might find something there. >>>> >> >>>> >> --- >>>> >> >>>> >> I don't know the Ericsson Language that well myself, but having had a >>>> look >>>> >> at it I decided to stay away from this technology. >>>> >> >>>> >> What I am doing is writing IVR's in Java, C# or even C++ and I >>>> decided to >>>> >> use vxml/ccxml to bring IVR capability into the standard dev >>>> environment >>>> >> so >>>> >> I can deal with business in a proper language. >>>> >> >>>> >> Jan >>>> >> >>>> >> -----Original Message----- >>>> >> From: freeswitch-users-bounces at lists.freeswitch.org >>>> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>>> David >>>> >> Swardstrom >>>> >> Sent: 24. juni 2010 17:52 >>>> >> To: freeswitch-users >>>> >> Subject: [Freeswitch-users] Erlang Examples >>>> >> >>>> >> I have been using JavaScript to handle a Conferencing application >>>> that >>>> >> started >>>> >> with the conf-ivr.js example program but is significantly more >>>> complex. >>>> >> This has been fun even though I had never used JavaScript before this >>>> >> year. >>>> >> >>>> >> However, there are things that seem to not be possible using >>>> JavaScript. >>>> >> I need to interact with several web based applications for several >>>> reasons >>>> >> and also need to provide some time based interactions with FreeSwitch >>>> >> and/or >>>> >> artifacts (Database entries, files of recorded conferences, etc). >>>> >> >>>> >> We (RemoteLink) have decided that the best solution for this support >>>> is >>>> >> to use an Erlang program and mod_erlang_event. So now I need to learn >>>> >> another language. >>>> >> >>>> >> But one thing that I do not find one the FreeSWITCH site is any >>>> Erlang >>>> >> examples. >>>> >> Are there some sample programs available such as one that would look >>>> for >>>> >> a certain type of event and print it out? >>>> >> >>>> >> I have found some semi-samples in the freeswitch-users archives but >>>> am >>>> >> somewhat >>>> >> ambivalent about using any of these without permission. >>>> >> >>>> >> Regards, >>>> >> David Swardstrom >>>> >> (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom >>>> >> >>>> >> _______________________________________________ >>>> >> FreeSWITCH-users mailing list >>>> >> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.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 >>>> > >>>> > >>>> > >>>> > -- >>>> > 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 >>>> > >>>> > _______________________________________________ >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> >>>> >>>> >>>> -- >>>> Blog: http://www.dujinfang.com >>>> Proj: http://www.freeswitch.org.cn >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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/20100630/f0d1de80/attachment-0001.html From shakad at gtt.co.gy Mon Jun 28 08:03:16 2010 From: shakad at gtt.co.gy (Shaka Dow) Date: Mon, 28 Jun 2010 11:03:16 -0400 (GMT-04:00) Subject: [Freeswitch-users] Can I add to directory without RELOADXML? In-Reply-To: <8005102.3541277737127656.JavaMail.SYSTEM@tel-mis-shaka> Message-ID: <9926220.3561277737392660.JavaMail.SYSTEM@tel-mis-shaka> Dear Freeswitch Users, I am a new member to the community and have been using Asterisk for years. I am very excited by what Freeswitch has to offer but in lieu of an equivalent to "Asterisk realtime", can someone let me know if the following is possible: - Can a new file be added to the internal directory and loaded without running the command "sofia profile internal rescan reloadxml" to simply load that one new file? If there are 100,000 files in /usr/local/freeswitch/conf/directory/default then the reload will take a few minutes. Surely there must be a better way to add "extensions" without executing such a processor intensive command since I will be adding users very frequently while in production. Is there a way to dynamically add to the DOM via the CLI or mod_event_socket? Looking forward to your feedback. Very best regards, Shaka Dow, Freeswitch User From mashudi72 at gmail.com Tue Jun 29 00:59:36 2010 From: mashudi72 at gmail.com (mashudi72 -) Date: Tue, 29 Jun 2010 14:59:36 +0700 Subject: [Freeswitch-users] Using fifo_orbit_announce For On-hook Agent In-Reply-To: References: Message-ID: Dear All, using FIFO, how to enable caller ID from caller? 2010/6/29 Michael Collins > I am not sure I understand what your setup is. Do you have a separate FIFO > queue for each agent? That would be the only way that each agent could have > his own specific announcement. Each FIFO queue has only one set of orbit > parameters (extension, context, dialplan, announcement file). What are you > trying to accomplish? > > -MC > > On Fri, Jun 25, 2010 at 6:39 AM, afshin afzali wrote: > >> Hi FreeSWITCH, >> >> I want to use fifo_orbit_announce to play specific agent greeting to his >> caller (can not insert it to caller dialplan). As my agents are on-hook >> (they use extensions to login / logout of queues) , I don't know if I could >> set this variable in login extension! If I can not use this way, is there >> any other way? >> >> appreciate all comments, >> BEST, >> -- afshin >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100629/267dbdc0/attachment-0001.html From mkane02 at harris.com Tue Jun 29 07:40:32 2010 From: mkane02 at harris.com (Kane, Michael (mkane02)) Date: Tue, 29 Jun 2010 10:40:32 -0400 Subject: [Freeswitch-users] Connect two FreeSWITCH Boxes Message-ID: Hello newbie here. I'm having difficulty routing calls between 2 FreeSWITCH servers. I'm following the instructions on the Wiki titled "Connect Two FreeSWITCH Boxes" and I'm taking the IP authentication approach. When I originate a call from server "B" the Regex matches and forwards the call onto Server "A". Server "A" isn't matching in dialplan/default.xml (Local_Call) for some reason. Public.xml successfully transfers the call into the default context, but for some reason it skips over the Local_Call Regex. Anyone's guidance would be greatly appreciated. Server "B" (call origination dialing 10001 from 20001) 10.3.3.10 dialplan/default.xml outbound call trace from console Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->Local_Call] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [Local_Call] destination_number(10001) =~ /^(20\d\d\d)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->Dial to BoxA] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (PASS) [Dial to BoxA] destination_number(10001) =~ /^(10\d\d\d)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 Action bridge(sofia/internal/10001 at 10.1.1.10) dialplan/public.xml ************************************************************************ ** Server "A" (Call termination) 10.1.1.10 dialplan/default.xml dialplan/public.xml inbound call trace from console Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [public->Calls from BoxB] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (PASS) [Calls from BoxB] destination_number(10001) =~ // break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 Action transfer($1 XML default) "there are a bunch of state machine messages here" Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->Local_Call] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [Local_Call] destination_number($1) =~ /^(10\d\d\d)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->Dial to BoxB] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [Dial to BoxB] destination_number($1) =~ /^(20\d\d\d)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->outbound] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [outbound] destination_number($1) =~ /^(\d)(\d{6})$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->pizza_demo] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [pizza_demo] destination_number($1) =~ /^(pizza|74992)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->local.example.com] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [local.example.com] ${toll_allow}() =~ /local/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->domestic.example.com] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [domestic.example.com] ${toll_allow}() =~ /domestic/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->international.example.com] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [international.example.com] ${toll_allow}() =~ /international/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->enum] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (PASS) [enum] ${module_exists(mod_enum)}(true) =~ /true/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (PASS) [enum] destination_number($1) =~ /^(.*)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 Action transfer($1 enum) I was expecting "Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [Local_Call] destination_number($1) =~ /^(10\d\d\d)$/ break=on-false" to match the dialed number for which it would route the call to 10001. By the way 10001, 10002 and 20001, 20002 have sucessfully registered with their respective servers and can place calls internally to the switch. Also when I run tcpdump on Server "B" I'm getting 404 Not Found, which indicates either the endpoint is not registered or properly configured. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100629/797e2b29/attachment-0001.html From jmesquita at freeswitch.org Tue Jun 29 21:50:10 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 30 Jun 2010 01:50:10 -0300 Subject: [Freeswitch-users] Can I add to directory without RELOADXML? In-Reply-To: <9926220.3561277737392660.JavaMail.SYSTEM@tel-mis-shaka> References: <8005102.3541277737127656.JavaMail.SYSTEM@tel-mis-shaka> <9926220.3561277737392660.JavaMail.SYSTEM@tel-mis-shaka> Message-ID: /JM On Mon, Jun 28, 2010 at 12:03 PM, Shaka Dow wrote: > Dear Freeswitch Users, > I am a new member to the community and have been using Asterisk for years. > I am very excited by what Freeswitch has to offer but in lieu of an > equivalent to "Asterisk realtime", can someone let me know if the following > is possible: > > - Can a new file be added to the internal directory and loaded without > running the command "sofia profile internal rescan reloadxml" to simply load > that one new file? > Yes, reloadxml alone will do that. That is without the sofia stuff and it won't drop calls or anything else. > > If there are 100,000 files in /usr/local/freeswitch/conf/directory/default > then the reload will take a few minutes. Surely there must be a better way > to add "extensions" without executing such a processor intensive command > since I will be adding users very frequently while in production. Is there a > way to dynamically add to the DOM via the CLI or mod_event_socket? > mod_xml_curl is your answer. Look here: http://wiki.freeswitch.org/wiki/Mod_xml_curl > > Looking forward to your feedback. > > Very best regards, > Shaka Dow, Freeswitch User > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/f179dc99/attachment.html From jonas.gauffin at gmail.com Tue Jun 29 22:20:33 2010 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 30 Jun 2010 07:20:33 +0200 Subject: [Freeswitch-users] Incompatible destination In-Reply-To: References: Message-ID: Ok. I found the RFC with all valid names (rfc3551, correct?). Shouldn't it work anyway since "8" is specified in the "m" attribute? (From RC4566: If an attribute is received that is not understood, it MUST be ignored by the receiver) //Jonas On Mon, Jun 28, 2010 at 8:23 PM, Brian West wrote: > Omission of the RTP map is fine if its a standard number like 0, 8, 18 and > such that are assigned. What isn't valid is G.729a or G.729b both are 100% > invalid. There is no such thing in the specs. > > /b > > On Jun 28, 2010, at 12:51 PM, Jonas Gauffin wrote: > > > FreeSwitch gives me INCOMPATIBLE_DESTINATION. My trunk provider says that > the SDP:s are valid. The first one looks OK by me. How about the second one? > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/03dd53ab/attachment.html From irmatov at gmail.com Wed Jun 30 00:02:18 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Wed, 30 Jun 2010 12:02:18 +0500 Subject: [Freeswitch-users] mod_erlang_event problem In-Reply-To: References: <20100624145749.GA17555@hijacked.us> <20100625150622.GE17555@hijacked.us> Message-ID: Ok, this is getting pretty confusing. >> So you use both mechanisms on a box with older freeswitch and erlang and >> both work fine? Can you narrow it down by matching versions on one side >> (freeswitch might be best) and see if the bug goes away/comes back? Its >> possible the new erlang releases have changed something in ei, but its >> more likely that I broke something in FreeSWITCH. What are the versions >> of FS/erlang on the box that works? > > Yes, both mechanisms work on other box. It is Debian Lenny, 64 bit. > Erlang is installed from Debian packages, it is 12B03. As for > freeswitch, it is a little bit difficult to tell its version. I was > installing it from svn repository, and can't tell exactly what > revision that was. May be 16041, but definitely before 17000. version > command reports just 'FreeSWITCH Version 1.0.trunk (hacked)'. I'll try > compiling 16041 on Red Hat machine and tell you the results. I can't get freeswitch and erlang play together even on old (working) server. It *has* a working installation of freeswitch and erlang (debian packages) that works. What I cannot do - is install another freeswitch on this same machine and make it work with either system provided erlang (R12B03) nor latest R13B04. I have tried newest freeswitch from git, and old revision 16041 from svn with both erlangs. Both exhibit exact same behaviour - call is disconnected, log file says 'got pid!' and 'exit erlang_outbound_session'. I am completely confused. Banging my head against the wall.. :) Any suggestions? -- Timur Irmatov, xmpp:irmatov at jabber.ru From steveayre at gmail.com Wed Jun 30 00:56:57 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 30 Jun 2010 08:56:57 +0100 Subject: [Freeswitch-users] Incompatible destination In-Reply-To: References: Message-ID: You won't be able to use G729 (it's either 18 or G729/8000 neither of which appear). The 8 (PCMA) should work IF you have PCMA enabled. Check which codecs you're allowing. Your FS might also be configured to leave the decision up to the destination if you're bridging, so check whether that endpoint supports PCMA too. And if you're doing PCMA-G729 that will fail with incompatibile destination too unless you're using the commercial (licensed) version of mod_com_g729. -Steve On 30 June 2010 06:20, Jonas Gauffin wrote: > Ok. I found the RFC with all valid names (rfc3551, correct?). > > Shouldn't it work anyway since "8" is specified in the "m" attribute? > (From RC4566: If an attribute is received that is not understood, it MUST > be ignored by the receiver) > > //Jonas > > On Mon, Jun 28, 2010 at 8:23 PM, Brian West wrote: > >> Omission of the RTP map is fine if its a standard number like 0, 8, 18 and >> such that are assigned. What isn't valid is G.729a or G.729b both are 100% >> invalid. There is no such thing in the specs. >> >> /b >> >> On Jun 28, 2010, at 12:51 PM, Jonas Gauffin wrote: >> >> > FreeSwitch gives me INCOMPATIBLE_DESTINATION. My trunk provider says >> that the SDP:s are valid. The first one looks OK by me. How about the second >> one? >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100630/ab4fbcdb/attachment.html From steveayre at gmail.com Wed Jun 30 00:58:44 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 30 Jun 2010 08:58:44 +0100 Subject: [Freeswitch-users] SIP header on only one fork of a bridge In-Reply-To: <4C28EDB8.6080609@communicatefreely.net> References: <4C28EDB8.6080609@communicatefreely.net> Message-ID: On 28 June 2010 19:45, Tim St. Pierre wrote: > Hello list, > > I would like to bridge a call to multiple SIP endpoints, but add different > headers to each. > > I'm not entirely sure how to do this. I have no problem exporting a SIP > header that does what I > want for one destination, but I'm not sure how to set it for two. > > My application is that I want two IP phones to ring - one with the internal > ring-ring splash, the > others with a group-answer (single ring, then lamp flash only), for > administrative assistants, etc. > > How do I export different variables to each branch? > > Thanks! > > -Tim > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/6cffd26a/attachment.html From jonas.gauffin at gmail.com Wed Jun 30 02:19:29 2010 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 30 Jun 2010 11:19:29 +0200 Subject: [Freeswitch-users] Incompatible destination In-Reply-To: References: Message-ID: both PMCU and PCMA are activated. On Wed, Jun 30, 2010 at 9:56 AM, Steven Ayre wrote: > You won't be able to use G729 (it's either 18 or G729/8000 neither of which > appear). > > The 8 (PCMA) should work IF you have PCMA enabled. Check which codecs > you're allowing. > > Your FS might also be configured to leave the decision up to the > destination if you're bridging, so check whether that endpoint supports PCMA > too. > > And if you're doing PCMA-G729 that will fail with incompatibile destination > too unless you're using the commercial (licensed) version of mod_com_g729. > > -Steve > > > > On 30 June 2010 06:20, Jonas Gauffin wrote: > >> Ok. I found the RFC with all valid names (rfc3551, correct?). >> >> Shouldn't it work anyway since "8" is specified in the "m" attribute? >> (From RC4566: If an attribute is received that is not understood, it MUST >> be ignored by the receiver) >> >> //Jonas >> >> On Mon, Jun 28, 2010 at 8:23 PM, Brian West wrote: >> >>> Omission of the RTP map is fine if its a standard number like 0, 8, 18 >>> and such that are assigned. What isn't valid is G.729a or G.729b both are >>> 100% invalid. There is no such thing in the specs. >>> >>> /b >>> >>> On Jun 28, 2010, at 12:51 PM, Jonas Gauffin wrote: >>> >>> > FreeSwitch gives me INCOMPATIBLE_DESTINATION. My trunk provider says >>> that the SDP:s are valid. The first one looks OK by me. How about the second >>> one? >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/097b061b/attachment-0001.html From helmut.kuper at ewetel.de Wed Jun 30 03:04:55 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 30 Jun 2010 12:04:55 +0200 Subject: [Freeswitch-users] ESL: Question about BACKGROUND_JOB events In-Reply-To: <21D9FC45-0FA9-4454-97AD-010E150470FF@avgs.ca> References: <4C22301C.3080405@ewetel.de> <4274374276864666200@unknownmsgid> <4C29BCF6.90701@ewetel.de> <21D9FC45-0FA9-4454-97AD-010E150470FF@avgs.ca> Message-ID: <4C2B16C7.6050200@ewetel.de> Hi, well, so I have to parse all BACKGROUND-JOB events in my app to find the event for me? On 29.06.2010 21:26, Mathieu Rene wrote: > Hrum.... [origination_uuid=xxx], it wont work in { } since its a per-leg param. > -- Mit freundlichen Gr??en Helmut Kuper Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Dr. Werner Brinker Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From steveayre at gmail.com Wed Jun 30 03:52:11 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 30 Jun 2010 11:52:11 +0100 Subject: [Freeswitch-users] Incompatible destination In-Reply-To: References: Message-ID: Then I suggest you pastebin your configuration, and debug-level freeswitch logs for the call, with sip tracing enabled so that we can see why PCMA isn't being used. http://pastebin:freeswitch at pastebin.freeswitch.org/ and send the link here when submitted. -Steve On 30 June 2010 10:19, Jonas Gauffin wrote: > both PMCU and PCMA are activated. > > > On Wed, Jun 30, 2010 at 9:56 AM, Steven Ayre wrote: > >> You won't be able to use G729 (it's either 18 or G729/8000 neither of >> which appear). >> >> The 8 (PCMA) should work IF you have PCMA enabled. Check which codecs >> you're allowing. >> >> Your FS might also be configured to leave the decision up to the >> destination if you're bridging, so check whether that endpoint supports PCMA >> too. >> >> And if you're doing PCMA-G729 that will fail with incompatibile >> destination too unless you're using the commercial (licensed) version of >> mod_com_g729. >> >> -Steve >> >> >> >> On 30 June 2010 06:20, Jonas Gauffin wrote: >> >>> Ok. I found the RFC with all valid names (rfc3551, correct?). >>> >>> Shouldn't it work anyway since "8" is specified in the "m" attribute? >>> (From RC4566: If an attribute is received that is not understood, it MUST >>> be ignored by the receiver) >>> >>> //Jonas >>> >>> On Mon, Jun 28, 2010 at 8:23 PM, Brian West wrote: >>> >>>> Omission of the RTP map is fine if its a standard number like 0, 8, 18 >>>> and such that are assigned. What isn't valid is G.729a or G.729b both are >>>> 100% invalid. There is no such thing in the specs. >>>> >>>> /b >>>> >>>> On Jun 28, 2010, at 12:51 PM, Jonas Gauffin wrote: >>>> >>>> > FreeSwitch gives me INCOMPATIBLE_DESTINATION. My trunk provider says >>>> that the SDP:s are valid. The first one looks OK by me. How about the second >>>> one? >>>> > >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100630/ef53628d/attachment.html From chaitanya at vivainfomedia.com Wed Jun 30 04:07:55 2010 From: chaitanya at vivainfomedia.com (Chaitanya Bhatt // Viva) Date: Wed, 30 Jun 2010 16:37:55 +0530 Subject: [Freeswitch-users] FreeSwitch Configuration : Help in SIP provider & gateway configuration In-Reply-To: References: Message-ID: Thanks for valuable response . I am trying using PRI on Sangoma Card. I am following this link http://wiki.sangoma.com/wanpipe-freeswitch-install to install wanpipe with freeswitch. But when i tried make install_pri i got following error : Checking for SCTP Utilities....OK. ./Setup: line 4850: ./get_sangoma_prid.sh: No such file or directory Failed to obtain SMG PRI package I am not getting how to resolve this issue. Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya at vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 On Fri, Jun 25, 2010 at 7:19 PM, Rupa Schomaker wrote: > I'm a little confused. You say that you will be running without SIP (cards > + PRI) but then ask about setting up SIP. Which configuration are you > trying to do? > > On Fri, Jun 25, 2010 at 2:12 AM, Chaitanya Bhatt // Viva < > chaitanya at vivainfomedia.com> wrote: > >> Hey >> >> I have installed FreeSwitch successfully but not getting how to use it. I >> want to use FreeSwitch in Inbound/Outbound IVRS application. >> We will be using FreeSwitch with Sangoma Card & PRIs. In configuration >> section i am not getting SIP provider & gateway configuration. >> I am newbie in FreeSwitch, Can you please guide how to proceed with SIP >> provider & gateway configuration ? >> >> Incase of any further queries, Please feel free to mail me or contact me >> on the numbers provided below. >> >> Thanks & Regards, >> Chaitanya Bhatt >> Software Engineer. >> >> Viva Infomedia Pvt. Ltd. >> 242, Oshiwara Industrial Centre, >> New Link Road, Opp. Oshiwara Bus Depot, >> Goregaon West, Mumbai 400104. >> >> Direct: +91.22.40310356 >> Board: +91.22.40310310 >> Email : chaitanya at vivainfomedia.com >> >> Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India >> Awards 2009 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/bac9812d/attachment.html From helmut.kuper at ewetel.de Wed Jun 30 04:33:40 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 30 Jun 2010 13:33:40 +0200 Subject: [Freeswitch-users] Question about PSTN support in FS Message-ID: <4C2B2B94.4020205@ewetel.de> Hello, I wonder about the fact that there are two PSTN modules for FS now: mod_ftdm and mod_openzap. I found this in FS wiki: "FreeTDM is the eventual replacement for OpenZAP. For now it is used to implement the boost PRI and BRI stacks for Sangoma cards." Is there already a clear decision what the "future safe" way is? regards Helmut From a.afzali2003 at gmail.com Wed Jun 30 04:38:02 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Wed, 30 Jun 2010 16:08:02 +0430 Subject: [Freeswitch-users] Using fifo_orbit_announce For On-hook Agent In-Reply-To: References: Message-ID: If your phone supports SIP update message, the actual caller-id will be updated when the agent respond to call. On 6/29/10, mashudi72 - wrote: > Dear All, > > using FIFO, how to enable caller ID from caller? > > > 2010/6/29 Michael Collins > >> I am not sure I understand what your setup is. Do you have a separate FIFO >> queue for each agent? That would be the only way that each agent could >> have >> his own specific announcement. Each FIFO queue has only one set of orbit >> parameters (extension, context, dialplan, announcement file). What are you >> trying to accomplish? >> >> -MC >> >> On Fri, Jun 25, 2010 at 6:39 AM, afshin afzali >> wrote: >> >>> Hi FreeSWITCH, >>> >>> I want to use fifo_orbit_announce to play specific agent greeting to his >>> caller (can not insert it to caller dialplan). As my agents are on-hook >>> (they use extensions to login / logout of queues) , I don't know if I >>> could >>> set this variable in login extension! If I can not use this way, is there >>> any other way? >>> >>> appreciate all comments, >>> BEST, >>> -- afshin >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > From peder at networkoblivion.com Wed Jun 30 04:47:25 2010 From: peder at networkoblivion.com (Peder) Date: Wed, 30 Jun 2010 06:47:25 -0500 Subject: [Freeswitch-users] Connect two FreeSWITCH Boxes In-Reply-To: References: Message-ID: <053901cb184a$02596290$070c27b0$@com> I would guess there is an error in public on A, but you didn't include all of that context so I can't say for sure. Note that it matches 10001 in public, but when it gets to Local_Call, it just sees $1, not 10001. That s why the match is failing. Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (PASS) [Calls from BoxB] destination_number(10001) =~ // break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 Action transfer($1 XML default) "there are a bunch of state machine messages here" Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->Local_Call] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [Local_Call] destination_number($1) =~ /^(10\d\d\d)$/ break=on-false From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kane, Michael (mkane02) Sent: Tuesday, June 29, 2010 9:41 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Connect two FreeSWITCH Boxes Hello newbie here. I'm having difficulty routing calls between 2 FreeSWITCH servers. I'm following the instructions on the Wiki titled "Connect Two FreeSWITCH Boxes" and I'm taking the IP authentication approach. When I originate a call from server "B" the Regex matches and forwards the call onto Server "A". Server "A" isn't matching in dialplan/default.xml (Local_Call) for some reason. Public.xml successfully transfers the call into the default context, but for some reason it skips over the Local_Call Regex. Anyone's guidance would be greatly appreciated. Server "B" (call origination dialing 10001 from 20001) 10.3.3.10 dialplan/default.xml outbound call trace from console Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->Local_Call] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [Local_Call] destination_number(10001) =~ /^(20\d\d\d)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->Dial to BoxA] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (PASS) [Dial to BoxA] destination_number(10001) =~ /^(10\d\d\d)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 Action bridge(sofia/internal/10001 at 10.1.1.10) dialplan/public.xml ************************************************************************** Server "A" (Call termination) 10.1.1.10 dialplan/default.xml dialplan/public.xml inbound call trace from console Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [public->Calls from BoxB] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (PASS) [Calls from BoxB] destination_number(10001) =~ // break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 Action transfer($1 XML default) "there are a bunch of state machine messages here" Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->Local_Call] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [Local_Call] destination_number($1) =~ /^(10\d\d\d)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->Dial to BoxB] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [Dial to BoxB] destination_number($1) =~ /^(20\d\d\d)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->outbound] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [outbound] destination_number($1) =~ /^(\d)(\d{6})$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->pizza_demo] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [pizza_demo] destination_number($1) =~ /^(pizza|74992)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->local.example.com] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [local.example.com] ${toll_allow}() =~ /local/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->domestic.example.com] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [domestic.example.com] ${toll_allow}() =~ /domestic/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->international.example.com] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [international.example.com] ${toll_allow}() =~ /international/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 parsing [default->enum] continue=false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (PASS) [enum] ${module_exists(mod_enum)}(true) =~ /true/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (PASS) [enum] destination_number($1) =~ /^(.*)$/ break=on-false Dialplan: sofia/internal/20001 at 10.3.3.10 Action transfer($1 enum) I was expecting "Dialplan: sofia/internal/20001 at 10.3.3.10 Regex (FAIL) [Local_Call] destination_number($1) =~ /^(10\d\d\d)$/ break=on-false" to match the dialed number for which it would route the call to 10001. By the way 10001, 10002 and 20001, 20002 have sucessfully registered with their respective servers and can place calls internally to the switch. Also when I run tcpdump on Server "B" I'm getting 404 Not Found, which indicates either the endpoint is not registered or properly configured. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/44888080/attachment-0001.html From gmaruzz at celliax.org Wed Jun 30 05:01:22 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 30 Jun 2010 14:01:22 +0200 Subject: [Freeswitch-users] Upgraded from 1.0.4 pre8 to the latest Git tree. Skype does not work anymore. In-Reply-To: <4C2A0CF5.4090606@gmail.com> References: <4C2A0CF5.4090606@gmail.com> Message-ID: On Tue, Jun 29, 2010 at 5:10 PM, Svetik wrote: > Hi, > > On weekend I have upgraded to the latest Git tree from 1.0.4 pre8 which > I was running for a long time, year may be. Everything went smooth, > except Skype does not work anymore. Basically I followed Download & Have you updated the configuration file? Maybe the configuration file format has changed from the one you are using... -giovanni > Installation Guide (http://wiki.freeswitch.org/wiki/Installation_Guide) > and Mod skypopen Skype Endpoint and Trunk > (http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk). > The only difference, I have added at the > end of the modules.conf.xml file to load skypopen module. > > My impression is that mod_skypopen hangs during initialization and > freeswitch or some of its parts are locked and do not respond properly. > When I call any number, I am getting busy signal immediately after > keying the number. > Log file shows: > 2010-06-28 18:58:40.334877 [CRIT] switch_core_session.c:1524 Throttle > Error! 0 > In my case freeswitch and skype are started automatically by the scripts > that are basically recommended ones. > > I have tried to load mod_skypopen manually (I removed mod_skypopen from > the modules.conf.xml file) and most of the time it hangs. > > 1. I start the computer, freeswitch and skype autostart, but freeswitch > does load mod_skypopen (it is not in the modules.conf.xml) > 2. I shutdown freeswitch and start it from the console and wait for its > prompt. > 3. I do load mod_skypopen from the prompt. > 4. I do load mod_skypopen from the prompt. > 5. At this time it never returns to the prompt and there is no ringtone, > log file shows: > 2010-06-28 18:16:45.296626 [NOTICE] switch_channel.c:776 New Channel > sofia/internal/1000 at 192.168.0.120 [d64d747e-8302-11df-91f3-1d2722671cfc] > > Occasionally, it works, it happens only the first time I am loading the > module after machine was rebooted, when I restart freeswitch after that > and try to load module again it always hangs, so I have to restart computer. > > When it hangs, freeswitch can not be shutdown with freeswitch -stop, I > have to actually kill freeswitch process in order to restart it. After > restarting freeswitch and Skype as well it follows the same bad scenario > and hangs. I have to restart computer, and after that sometimes it works. > > Scripts to manage skype and freeswitch work perfect with version 1.0.4 > pre8, I can restart freeswitch and skype multiple time no problem. > I still have 1.0.4 pre8 sitting around and if I switch to it, it works > perfectly. > > I am running recommended version of Skype 2.0.72 > Everything is run on the 1Ghz machine (512MB RAM) > > Any thoughts what is wrong with my configuration? > Anything I can try to resolve this issue? > > Thank you, > > Igor > > Below are samples of logs for "bad" and "good" loads. > > ----------------------------------- bad load > ------------------------------------------- > freeswitch at pbx> load mod_sk2010-06-27 20:40:52.462365 [WARNING] > sofia.c:3700 Ping succeeded voipms with code 200 - count -1/1/1, state UP > ypopen > 2010-06-27 20:41:01.721128 [DEBUG] mod_skypopen.c:1223 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1223 > ][none ? ? ?][-1,-1,-1] globals.debug=8 > 2010-06-27 20:41:01.724336 [DEBUG] mod_skypopen.c:1230 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1230 > ][none ? ? ?][-1,-1,-1] globals.dialplan=XML > 2010-06-27 20:41:01.724336 [DEBUG] mod_skypopen.c:1227 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1227 > ][none ? ? ?][-1,-1,-1] globals.context=default > 2010-06-27 20:41:01.724336 [DEBUG] mod_skypopen.c:1233 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1233 > ][none ? ? ?][-1,-1,-1] globals.destination=5000 > 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1236 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1236 > ][none ? ? ?][-1,-1,-1] globals.skype_user=Boltik > 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1239 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1239 > ][none ? ? ?][-1,-1,-1] globals.report_incoming_chatmessages=true > 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1242 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1242 > ][none ? ? ?][-1,-1,-1] globals.silent_mode=false > 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1245 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1245 > ][none ? ? ?][-1,-1,-1] globals.write_silence_when_idle=true > 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1341 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1341 > ][none ? ? ?][-1,-1,-1] interface_id=1 > 2010-06-27 20:41:01.725354 [DEBUG] mod_skypopen.c:1352 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1352 > ][none ? ? ?][-1,-1,-1] name=Boltik > 2010-06-27 20:41:01.726369 [DEBUG] mod_skypopen.c:1358 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1358 > ][none ? ? ?][-1,-1,-1] Initialized XInitThreads! > 2010-06-27 20:41:01.727392 [DEBUG] mod_skypopen.c:1381 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1381 > ][Boltik ?][-1, 0, 0] CONFIGURING interface_id=1 > 2010-06-27 20:41:01.727392 [DEBUG] mod_skypopen.c:1418 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1418 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].name=Boltik > 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1421 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1421 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].context=default > 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1424 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1424 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].dialplan=XML > 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1427 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1427 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].destination=5000 > 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1430 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1430 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].X11_display=:101 > 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1433 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1433 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].skype_user=Boltik > 2010-06-27 20:41:01.728480 [DEBUG] mod_skypopen.c:1436 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1436 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].report_incoming_chatmessages=1 > 2010-06-27 20:41:01.729539 [DEBUG] mod_skypopen.c:1439 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1439 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].silent_mode=0 > 2010-06-27 20:41:01.729539 [DEBUG] mod_skypopen.c:1442 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1442 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].write_silence_when_idle=1 > 2010-06-27 20:41:01.729539 [DEBUG] mod_skypopen.c:1445 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1445 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].setsockopt=0 > 2010-06-27 20:41:01.729539 [WARNING] mod_skypopen.c:1447 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][WARNINGA ?1447 > ][Boltik ?][-1, 0, 0] STARTING interface_id=1 > 2010-06-27 20:41:01.730932 [DEBUG] skypopen_protocol.c:1594 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1594 > ][Boltik ?][-1, 0, 0] X Display ':101' opened > 2010-06-27 20:41:01.730932 [DEBUG] skypopen_protocol.c:1536 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1536 > ][none ? ? ?][-1,-1,-1] Skype instance found with id #2097368 > 2010-06-27 20:41:01.833832 [DEBUG] mod_skypopen.c:1150 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1150 > ][Boltik ?][-1, 0, 0] In skypopen_signaling_thread_func: started, > p=0xb5bbaab8 > 2010-06-27 20:41:01.834965 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||OK||| > 2010-06-27 20:41:01.834965 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||PROTOCOL 7||| > 2010-06-27 20:41:01.834965 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| > 2010-06-27 20:41:01.834965 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||CURRENTUSERHANDLE Boltik||| > 2010-06-27 20:41:01.834965 [DEBUG] skypopen_protocol.c:263 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?263 > ][Boltik ?][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, > currentuserhandle: CURRENTUSERHANDLE, cuh: Boltik, skype_user: Boltik! > 2010-06-27 20:41:01.835975 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||USERSTATUS ONLINE||| > 2010-06-27 20:41:01.936026 [NOTICE] mod_skypopen.c:1472 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][NOTICA ?1472 > ][Boltik ?][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype > client and connect to its SKYPE API for interface_id=1 > 2010-06-27 20:41:01.936026 [NOTICE] mod_skypopen.c:1481 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][NOTICA ?1481 > ][Boltik ?][-1, 0, 0] Found a running Skype client, connected to its > SKYPE API for interface_id=1, waiting 60 seconds for > CURRENTUSERHANDLE==Boltik > 2010-06-27 20:41:01.936026 [WARNING] mod_skypopen.c:1500 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][WARNINGA ?1500 > ][Boltik ?][-1, 0, 0] Interface_id=1 is now STARTED, the Skype client to > which we are connected gave us the correct CURRENTUSERHANDLE (Boltik) > 2010-06-27 20:41:01.937122 [DEBUG] skypopen_protocol.c:1494 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1494 > ][Boltik ?][-1, 0, 0] SENDING: |||PROTOCOL 7|||| > 2010-06-27 20:41:01.938438 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||PROTOCOL 7||| > 2010-06-27 20:41:02.006171 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| > 2010-06-27 20:41:02.008363 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||CURRENTUSERHANDLE Boltik||| > 2010-06-27 20:41:02.008363 [DEBUG] skypopen_protocol.c:263 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?263 > ][Boltik ?][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, > currentuserhandle: CURRENTUSERHANDLE, cuh: Boltik, skype_user: Boltik! > 2010-06-27 20:41:02.009489 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||USERSTATUS ONLINE||| > > ---------------------------------------------------------------------------------------- > > -------------------------------- good load > --------------------------------------- > freeswitch at pbx>load mod_skypopen > 2010-06-27 22:48:30.592753 [DEBUG] mod_skypopen.c:1223 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1223 > ][none ? ? ?][-1,-1,-1] globals.debug=8 > 2010-06-27 22:48:30.597741 [DEBUG] mod_skypopen.c:1230 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1230 > ][none ? ? ?][-1,-1,-1] globals.dialplan=XML > 2010-06-27 22:48:30.597741 [DEBUG] mod_skypopen.c:1227 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1227 > ][none ? ? ?][-1,-1,-1] globals.context=default > 2010-06-27 22:48:30.597741 [DEBUG] mod_skypopen.c:1233 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1233 > ][none ? ? ?][-1,-1,-1] globals.destination=5000 > 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1236 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1236 > ][none ? ? ?][-1,-1,-1] globals.skype_user=Boltik > 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1239 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1239 > ][none ? ? ?][-1,-1,-1] globals.report_incoming_chatmessages=true > 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1242 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1242 > ][none ? ? ?][-1,-1,-1] globals.silent_mode=false > 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1245 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1245 > ][none ? ? ?][-1,-1,-1] globals.write_silence_when_idle=true > 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1341 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1341 > ][none ? ? ?][-1,-1,-1] interface_id=1 > 2010-06-27 22:48:30.598753 [DEBUG] mod_skypopen.c:1352 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1352 > ][none ? ? ?][-1,-1,-1] name=Boltik > 2010-06-27 22:48:30.599763 [DEBUG] mod_skypopen.c:1358 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1358 > ][none ? ? ?][-1,-1,-1] Initialized XInitThreads! > 2010-06-27 22:48:30.600769 [DEBUG] mod_skypopen.c:1381 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1381 > ][Boltik ?][-1, 0, 0] CONFIGURING interface_id=1 > 2010-06-27 22:48:30.600769 [DEBUG] mod_skypopen.c:1418 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1418 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].name=Boltik > 2010-06-27 22:48:30.601785 [DEBUG] mod_skypopen.c:1421 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1421 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].context=default > 2010-06-27 22:48:30.601785 [DEBUG] mod_skypopen.c:1424 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1424 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].dialplan=XML > 2010-06-27 22:48:30.609658 [DEBUG] mod_skypopen.c:1427 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1427 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].destination=5000 > 2010-06-27 22:48:30.609658 [DEBUG] mod_skypopen.c:1430 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1430 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].X11_display=:101 > 2010-06-27 22:48:30.609658 [DEBUG] mod_skypopen.c:1433 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1433 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].skype_user=Boltik > 2010-06-27 22:48:30.609658 [DEBUG] mod_skypopen.c:1436 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1436 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].report_incoming_chatmessages=1 > 2010-06-27 22:48:30.610668 [DEBUG] mod_skypopen.c:1439 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1439 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].silent_mode=0 > 2010-06-27 22:48:30.610668 [DEBUG] mod_skypopen.c:1442 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1442 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].write_silence_when_idle=1 > 2010-06-27 22:48:30.610668 [DEBUG] mod_skypopen.c:1445 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1445 > ][Boltik ?][-1, 0, 0] interface_id=1 > globals.SKYPOPEN_INTERFACES[interface_id].setsockopt=0 > 2010-06-27 22:48:30.610668 [WARNING] mod_skypopen.c:1447 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][WARNINGA ?1447 > ][Boltik ?][-1, 0, 0] STARTING interface_id=1 > 2010-06-27 22:48:30.612003 [DEBUG] skypopen_protocol.c:1594 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1594 > ][Boltik ?][-1, 0, 0] X Display ':101' opened > 2010-06-27 22:48:30.613019 [DEBUG] skypopen_protocol.c:1536 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1536 > ][none ? ? ?][-1,-1,-1] Skype instance found with id #2097368 > 2010-06-27 22:48:30.716767 [DEBUG] mod_skypopen.c:1150 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1150 > ][Boltik ?][-1, 0, 0] In skypopen_signaling_thread_func: started, > p=0xb5bb7ab8 > 2010-06-27 22:48:30.716767 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||OK||| > 2010-06-27 22:48:30.717802 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||PROTOCOL 7||| > 2010-06-27 22:48:30.717802 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| > 2010-06-27 22:48:30.717802 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||CURRENTUSERHANDLE Boltik||| > 2010-06-27 22:48:30.717802 [DEBUG] skypopen_protocol.c:263 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?263 > ][Boltik ?][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, > currentuserhandle: CURRENTUSERHANDLE, cuh: Boltik, skype_user: Boltik! > 2010-06-27 22:48:30.717802 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||USERSTATUS NA||| > 2010-06-27 22:48:30.817902 [NOTICE] mod_skypopen.c:1472 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][NOTICA ?1472 > ][Boltik ?][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype > client and connect to its SKYPE API for interface_id=1 > 2010-06-27 22:48:30.817902 [NOTICE] mod_skypopen.c:1481 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][NOTICA ?1481 > ][Boltik ?][-1, 0, 0] Found a running Skype client, connected to its > SKYPE API for interface_id=1, waiting 60 seconds for > CURRENTUSERHANDLE==Boltik > 2010-06-27 22:48:30.817902 [WARNING] mod_skypopen.c:1500 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][WARNINGA ?1500 > ][Boltik ?][-1, 0, 0] Interface_id=1 is now STARTED, the Skype client to > which we are connected gave us the correct CURRENTUSERHANDLE (Boltik) > 2010-06-27 22:48:30.817902 [DEBUG] skypopen_protocol.c:1494 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1494 > ][Boltik ?][-1, 0, 0] SENDING: |||PROTOCOL 7|||| > 2010-06-27 22:48:30.817902 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||PROTOCOL 7||| > 2010-06-27 22:48:30.829711 [DEBUG] skypopen_protocol.c:1494 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1494 > ][Boltik ?][-1, 0, 0] SENDING: |||SET AUTOAWAY OFF|||| > 2010-06-27 22:48:30.829711 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||||| > 2010-06-27 22:48:30.840990 [DEBUG] skypopen_protocol.c:1494 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1494 > ][Boltik ?][-1, 0, 0] SENDING: |||SET WINDOWSTATE HIDDEN|||| > 2010-06-27 22:48:30.840990 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||WINDOWSTATE HIDDEN||| > 2010-06-27 22:48:30.897291 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| > 2010-06-27 22:48:30.901487 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||CURRENTUSERHANDLE Boltik||| > 2010-06-27 22:48:30.901487 [DEBUG] skypopen_protocol.c:263 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?263 > ][Boltik ?][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, > currentuserhandle: CURRENTUSERHANDLE, cuh: Boltik, skype_user: Boltik! > 2010-06-27 22:48:30.902499 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||USERSTATUS NA||| > 2010-06-27 22:48:30.903510 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||AUTOAWAY OFF||| > 2010-06-27 22:48:30.905503 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||USER Boltik ONLINESTATUS ONLINE||| > 2010-06-27 22:48:30.906524 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||USERSTATUS ONLINE||| > 2010-06-27 22:48:30.906524 [DEBUG] skypopen_protocol.c:1494 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1494 > ][Boltik ?][-1, 0, 0] SENDING: |||SET USERSTATUS ONLINE|||| > 2010-06-27 22:48:30.908668 [DEBUG] skypopen_protocol.c:176 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?176 > ][Boltik ?][-1, 0, 0] READING: |||USERSTATUS ONLINE||| > > +OK > > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1544 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1544 > ][Boltik ?][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].interface_id=1 > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1545 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1545 > ][Boltik ?][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].name=Boltik > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1546 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1546 > ][Boltik ?][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].context=default > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1547 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1547 > ][Boltik ?][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].dialplan=XML > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1548 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1548 > ][Boltik ?][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].destination=5000 > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1549 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1549 > ][Boltik ?][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].X11_display=:101 > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1550 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1550 > ][Boltik ?][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].skype_user=Boltik > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1552 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1552 > ][Boltik ?][-1, 0, 0] i=1 > globals.SKYPOPEN_INTERFACES[1].report_incoming_chatmessages=1 > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1553 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1553 > ][Boltik ?][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].silent_mode=0 > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1554 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1554 > ][Boltik ?][-1, 0, 0] i=1 > globals.SKYPOPEN_INTERFACES[1].write_silence_when_idle=1 > 2010-06-27 22:48:30.916757 [DEBUG] mod_skypopen.c:1555 rev > git2svn-syncpoint-master-121-g4e82098[(nil)|37 ? ? ][DEBUG_SKYPE ?1555 > ][Boltik ?][-1, 0, 0] i=1 globals.SKYPOPEN_INTERFACES[1].setsockopt=0 > 2010-06-27 22:48:30.916757 [CONSOLE] switch_loadable_module.c:944 > Successfully Loaded [mod_skypopen] > 2010-06-27 22:48:30.916757 [NOTICE] switch_loadable_module.c:145 Adding > Endpoint 'skypopen' > 2010-06-27 22:48:30.916757 [NOTICE] switch_loadable_module.c:273 Adding > API Function 'sk' > 2010-06-27 22:48:30.916757 [NOTICE] switch_loadable_module.c:273 Adding > API Function 'skypopen' > 2010-06-27 22:48:30.916757 [NOTICE] switch_loadable_module.c:273 Adding > API Function 'skypopen_chat' > 2010-06-27 22:48:30.916757 [NOTICE] switch_loadable_module.c:378 Adding > Chat interface 'skype' > freeswitch at pbx> > > ---------------------------------------------------------------------------------------- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From irmatov at gmail.com Wed Jun 30 05:21:30 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Wed, 30 Jun 2010 17:21:30 +0500 Subject: [Freeswitch-users] mod_erlang_event problem In-Reply-To: References: <20100624145749.GA17555@hijacked.us> <20100625150622.GE17555@hijacked.us> Message-ID: On Wed, Jun 30, 2010 at 12:02 PM, Timur Irmatov wrote: > Ok, this is getting pretty confusing. It seems that my problem was related to host name part was not properly configured in /etc/hosts. After fixing that *and* restarting epmd, I was able to connect latest freeswitch and erlang. -- Timur Irmatov, xmpp:irmatov at jabber.ru From steveayre at gmail.com Wed Jun 30 05:42:36 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 30 Jun 2010 13:42:36 +0100 Subject: [Freeswitch-users] Question about PSTN support in FS In-Reply-To: <4C2B2B94.4020205@ewetel.de> References: <4C2B2B94.4020205@ewetel.de> Message-ID: FreeTDM is the newer, so if it works ok for you now I'd say that's the "future safe" way. Assuming the module usage doesn't change. -Steve On 30 June 2010 12:33, Helmut Kuper wrote: > Hello, > > I wonder about the fact that there are two PSTN modules for FS now: > mod_ftdm and mod_openzap. > > I found this in FS wiki: > > "FreeTDM is the eventual replacement for OpenZAP. For now it is used to > implement the boost PRI and BRI stacks for Sangoma cards." > > > Is there already a clear decision what the "future safe" way is? > > > regards > Helmut > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/bfbebc4e/attachment.html From brian at freeswitch.org Wed Jun 30 05:59:25 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Jun 2010 07:59:25 -0500 Subject: [Freeswitch-users] ESL: Question about BACKGROUND_JOB events In-Reply-To: <4C2B16C7.6050200@ewetel.de> References: <4C22301C.3080405@ewetel.de> <4274374276864666200@unknownmsgid> <4C29BCF6.90701@ewetel.de> <21D9FC45-0FA9-4454-97AD-010E150470FF@avgs.ca> <4C2B16C7.6050200@ewetel.de> Message-ID: <3208B925-C778-4A6D-83F6-2CA836B6C365@freeswitch.org> You act like that is a hard task to accomplish? /b On Jun 30, 2010, at 5:04 AM, Helmut Kuper wrote: > Hi, > > well, so I have to parse all BACKGROUND-JOB events in my app to find the > event for me? > > > On 29.06.2010 21:26, Mathieu Rene wrote: >> Hrum.... [origination_uuid=xxx], it wont work in { } since its a per-leg param. >> > From mcampbellsmith at gmail.com Wed Jun 30 06:07:32 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 30 Jun 2010 23:07:32 +1000 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> Message-ID: Updating the configuration did not help. I'm not telling FS to use any other codecs other than what I have specified below, so I'm not sure what I have done wrong. Below is the full trace: ------------------------------------------------------------------------ INVITE sip:1020 at 192.168.1.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae From: 1000 ;tag=2c7a518d12f9370eo0 To: Call-ID: 316156b9-f6b413d2 at 192.168.1.121 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="1000",realm="192.168.1.120",nonce="77836a7a-8447-11df-93a6-d9ad5b204ca2",uri="sip:1020 at 192.168.1.120",algorithm=MD5,response="c4a5e5cb60676366b8bcfb1329f3fc08",qop=auth,nc=00000001,cnonce="6a0220d0" Contact: 1000 Expires: 240 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 451 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 17789112 17789112 IN IP4 192.168.1.121 s=- c=IN IP4 192.168.1.121 t=0 0 m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 343 bytes to udp/[192.168.1.121]:5060 at 13:00:32.882684: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae From: 1000 ;tag=2c7a518d12f9370eo0 To: Call-ID: 316156b9-f6b413d2 at 192.168.1.121 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19 14-49-15 -0500 Content-Length: 0 ------------------------------------------------------------------------ 2010-06-30 23:00:32.899423 [DEBUG] sofia.c:5975 IP 192.168.1.121 Rejected by acl "domains". Falling back to Digest auth. 2010-06-30 23:00:32.914917 [NOTICE] switch_channel.c:776 New Channel sofia/internal/1000 at 192.168.1.120 [77a3ab82-8447-11df-93a7-d9ad5b204ca2] 2010-06-30 23:00:32.943135 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.1.120) Running State Change CS_NEW 2010-06-30 23:00:32.945232 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000 at 192.168.1.120) State NEW 2010-06-30 23:00:32.989778 [DEBUG] sofia.c:4293 Channel sofia/internal/1000 at 192.168.1.120 entering state [received][100] 2010-06-30 23:00:32.997500 [DEBUG] sofia.c:4304 Remote SDP: v=0 o=- 17789112 17789112 IN IP4 192.168.1.121 s=- c=IN IP4 192.168.1.121 t=0 0 m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2010-06-30 23:00:33.002369 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:30]/[G729:18:8000:20] 2010-06-30 23:00:33.005967 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:30]/[PCMU:0:8000:20] 2010-06-30 23:00:33.008248 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:30]/[GSM:3:8000:20] 2010-06-30 23:00:33.010721 [DEBUG] sofia_glue.c:3924 Substituting codec PCMU at 30i@8000h 2010-06-30 23:00:33.019002 [DEBUG] sofia_glue.c:2462 Set Codec sofia/internal/1000 at 192.168.1.120 PCMU/8000 30 ms 240 samples 2010-06-30 23:00:33.038714 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf send/recv payload to 101 2010-06-30 23:00:33.040992 [DEBUG] sofia.c:4451 (sofia/internal/1000 at 192.168.1.120) State Change CS_NEW -> CS_INIT 2010-06-30 23:00:33.043182 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.1.120 [BREAK] 2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.1.120) Running State Change CS_INIT 2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.1.120) State INIT 2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:83 sofia/internal/1000 at 192.168.1.120 SOFIA INIT 2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:117 (sofia/internal/1000 at 192.168.1.120) State Change CS_INIT -> CS_ROUTING 2010-06-30 23:00:33.048752 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.1.120 [BREAK] 2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.1.120) State INIT going to sleep 2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.1.120) Running State Change CS_ROUTING 2010-06-30 23:00:33.048752 [DEBUG] switch_channel.c:1474 (sofia/internal/1000 at 192.168.1.120) Callstate Change DOWN -> RINGING 2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.1.120) State ROUTING 2010-06-30 23:00:33.055456 [DEBUG] switch_channel.c:1333 (sofia/internal/1000 at 192.168.1.120) Callstate Change RINGING -> ACTIVE 2010-06-30 23:00:33.055456 [DEBUG] mod_sofia.c:140 sofia/internal/1000 at 192.168.1.120 SOFIA ROUTING 2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000 at 192.168.1.120 Standard ROUTING 2010-06-30 23:00:33.055456 [INFO] mod_dialplan_xml.c:331 Processing 1000->1020 in context default Dialplan: sofia/internal/1000 at 192.168.1.120 parsing [default->Local_1000_1019] continue=false Dialplan: sofia/internal/1000 at 192.168.1.120 Regex (FAIL) [Local_1000_1019] destination_number(1020) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.1.120 parsing [default->Mobile_1020s] continue=false Dialplan: sofia/internal/1000 at 192.168.1.120 Regex (PASS) [Mobile_1020s] destination_number(1020) =~ /^(102[0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.1.120 Action set(dialed_extension=1020) Dialplan: sofia/internal/1000 at 192.168.1.120 Action export(codec_string=GSM) Dialplan: sofia/internal/1000 at 192.168.1.120 Action set(codec_string=GSM) Dialplan: sofia/internal/1000 at 192.168.1.120 Action bridge(user/${dialed_extension}@${domain}) Dialplan: sofia/internal/1000 at 192.168.1.120 Action set_user(1000@${domain}) Dialplan: sofia/internal/1000 at 192.168.1.120 Action answer() Dialplan: sofia/internal/1000 at 192.168.1.120 Action sleep(1000) Dialplan: sofia/internal/1000 at 192.168.1.120 Action system(/usr/local/freeswitch/scripts/sms.pl ${smsaccount} ${smspassword} ${smsnumber} 'You have one new voicemail from ${effec$ CS_EXECUTE 2010-06-30 23:00:33.114917 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.1.120 [BREAK] 2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.1.120) State ROUTING going to sleep 2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.1.120) Running State Change CS_EXECUTE 2010-06-30 23:00:33.128360 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 192.168.1.120) State EXECUTE 2010-06-30 23:00:33.129661 [DEBUG] mod_sofia.c:233 sofia/internal/1000 at 192.168.1.120 SOFIA EXECUTE 2010-06-30 23:00:33.134566 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000 at 192.168.1.120 Standard EXECUTE EXECUTE sofia/internal/1000 at 192.168.1.120 set(dialed_extension=1020) 2010-06-30 23:00:33.157698 [DEBUG] mod_dptools.c:843 sofia/internal/1000 at 192.168.1.120 SET [dialed_extension]=[1020] EXECUTE sofia/internal/1000 at 192.168.1.120 bridge(user/1020 at mydns.dyndns.org) 2010-06-30 23:00:33.248458 [DEBUG] switch_ivr_originate.c:1956 variable string 0 = [presence_id=1020 at mydns.dyndns.org] 2010-06-30 23:00:33.260573 [NOTICE] switch_channel.c:776 New Channel sofia/internal/sip:1020 at 192.168.1.123:5060 [77d7e3b6-8447-11df-93a8-d9ad5b204ca2] 2010-06-30 23:00:33.303487 [DEBUG] mod_sofia.c:3883 (sofia/internal/sip:1020 at 192.168.1.123:5060) State Change CS_NEW -> CS_INIT 2010-06-30 23:00:33.304564 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/sip:1020 at 192.168.1.123:5060 [BREAK] 2010-06-30 23:00:33.327507 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1020 at 192.168.1.123:5060) Running State Change CS_INIT 2010-06-30 23:00:33.332010 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:1020 at 192.168.1.123:5060) State INIT 2010-06-30 23:00:33.334621 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1020 at 192.168.1.123:5060 SOFIA INIT send 1183 bytes to udp/[192.168.1.123]:5060 at 13:00:33.351944: ------------------------------------------------------------------------ INVITE sip:1020 at 192.168.1.123:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bKFDpKe5v6j2QcQ Max-Forwards: 69 From: "1000" ;tag=Z51Qve55SUHta To: Call-ID: 4f442091-feea-122d-448a-00e04c0312e9 CSeq: 132833080 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19 14-49-15 -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 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display Remote-Party-ID: "1000" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1277875447 1277875448 IN IP4 192.168.1.120 s=FreeSWITCH c=IN IP4 192.168.1.120 t=0 0 m=audio 27386 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30 It fails after this with Not Acceptable Here / INCOMPATIBLE_DESTINATION On Mon, Jun 28, 2010 at 5:15 PM, David Ponzone wrote: > Please, retry with a genuine config (the default one would be a wise > choice). > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 28/06/2010 ? 07:37, Mark Campbell-Smith a ?crit : > > Hi All, > > I'm not really sure if I got a firm answer for this one. ?Is the only > way to ensure that transcoding is performed is by using > late-negotiation? ?Why aren't all my codecs sent in the INVITE message > to the B-leg (extension 1020)? > > Thanks! > > On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith > wrote: > > FS version and codecs are shown below, but my config file are probably > > quite old. ?But I guess they should still work? > > All codecs are loaded, and the call works if late negotiation is set > > on profile internal. > > As I wrote above: > > The call setup is extension 1000 calls extension 1020 > > 1. Extension 1000 calls with preferred codec PCMU. ?PCMU is chosen by > > FS as the A-leg codec > > 2. Extension 1020 only supports GSM codec. ?The call fails with Not > > Acceptable Here. > > I forgot to write that Extension 1000 does not support GSM (I want to > > force transcoding). ?Is that why FS is filtering out GSM on the b-leg? > > freeswitch at internal> version > > FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500) > > freeswitch at internal> show codecs > > type,name,ikey > > codec,ADPCM (IMA),mod_voipcodecs > > codec,G.711 alaw,CORE_PCM_MODULE > > codec,G.711 ulaw,CORE_PCM_MODULE > > codec,G.722,mod_voipcodecs > > codec,G.723.1 6.3k,mod_g723_1 > > codec,G.726 16k,mod_voipcodecs > > codec,G.726 16k (AAL2),mod_voipcodecs > > codec,G.726 24k,mod_voipcodecs > > codec,G.726 24k (AAL2),mod_voipcodecs > > codec,G.726 32k,mod_voipcodecs > > codec,G.726 32k (AAL2),mod_voipcodecs > > codec,G.726 40k,mod_voipcodecs > > codec,G.726 40k (AAL2),mod_voipcodecs > > codec,G.729,mod_com_g729 > > codec,GSM,mod_voipcodecs > > codec,H.261 Video (passthru),mod_h26x > > codec,H.263 Video (passthru),mod_h26x > > codec,H.263+ Video (passthru),mod_h26x > > codec,H.263++ Video (passthru),mod_h26x > > codec,H.264 Video (passthru),mod_h26x > > codec,LPC-10,mod_voipcodecs > > 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,Speex,mod_speex > > 25 total. > > > On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone > wrote: > > Mark, > > I confirm that, as I wrote that wiki page (the early negotiation part) :) > > Can you really confirm your FS version ? > > The parameter you showed is old. > > codec-prefs has been replaced in SIP profiles by: > > ?? ? > > ?? ? > > David Ponzone ?Direction Technique > > email: david.ponzone at ipeva.fr > > tel: ? ? ?01 74 03 18 97 > > gsm: ? 06 66 98 76 34 > > Service Client?IPeva > > tel: ? ? ?0811 46 26 26 > > www.ipeva.fr? -? ?www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > > non autoris?e est interdite. Tout message ?lectronique est susceptible > > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 23/06/2010 ? 14:43, Mark Campbell-Smith a ?crit : > > Check this good wiki page for how FS negotiates codecs (early > > negotiation default): > > http://wiki.freeswitch.org/wiki/Codec_Negotiation > > I have this set in my internal profile: > > ??? > > and as stated before vars.xml: > > > > > > Setting late negotiation works (thanks Sergey), but reading the wiki > > page, I see the following sentence, which I interpret that GSM should > > still be sent: > > When FS calls leg B, the list of codecs in outbound-codec-prefs for > > the SIP profile is reorganized by pushing the codec negotiated above > > for leg A at the top . If B does not accept any of the codecs, the > > calls fails, obviously. > > > > On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano > > wrote: > > I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? > > Wouldn't fs only get involved when its media server is referred to? > > If the "other endpoint" will only accept G729, doesn't that mean you > > need to change that endpoint to also accept G711 or also license G729 > > in FS? > > On 6/23/10, Mark Campbell-Smith wrote: > > Test Setup: > > vars.xml: > > ? > > ? > > The call setup is extension 1000 calls extension 1020 > > 1. Extension 1000 calls with preferred codec PCMU. ?PCMU is chosen by > > FS as the A-leg codec > > 2. Extension 1020 only supports GSM codec. ?The call fails with Not > > Acceptable Here. > > FS only offers G729 and PCMU to 1020. ?How do I change the number of > > codecs that are offered to an extension? ?I know I can change the > > order in the codec_prefs, but would prefer FS to offer all three > > codecs to an extension. > > ? ?m=audio 23662 RTP/AVP 0 18 101 13 > > ? ?a=rtpmap:0 PCMU/8000 > > ? ?a=rtpmap:18 G729/8000 > > ? ?a=rtpmap:101 telephone-event/8000 > > ? ?a=fmtp:101 0-16 > > ? ?a=rtpmap:13 CN/8000 > > ? ?a=ptime:30 > > Thanks > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > -- > > Sent from my mobile device > > ====================== > > Tony Graziano, Manager > > Telephone: 434.984.8430 > > sip: tgraziano at voice.myitdepartment.net > > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > > Telephone: 434.984.8426 > > sip: helpdesk at voice.myitdepartment.net > > Fax: 434.984.8427 > > Helpdesk Contract Customers: > > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > > Because 31 Oct = 25 Dec. > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 deya787 at gmail.com Wed Jun 30 06:07:38 2010 From: deya787 at gmail.com (Deya M) Date: Wed, 30 Jun 2010 16:07:38 +0300 Subject: [Freeswitch-users] Possible Mem LeakFS Message-ID: Hi, Starting with total memory usage of 300k, I go up to 2GB after 10 hours running FS. Just doing further tests, I got the log file, which at the end has: ==4573== LEAK SUMMARY: ==4573== definitely lost: 8,166 bytes in 26 blocks ==4573== indirectly lost: 284 bytes in 1 blocks ==4573== possibly lost: 2,794,680 bytes in 477 blocks ==4573== still reachable: 74,666 bytes in 2,057 blocks ==4573== suppressed: 0 bytes in 0 blocks ==4573== ==4573== ERROR SUMMARY: 232006 errors from 122 contexts (suppressed: 326 from 13) Appreciate your help! http://pastebin.freeswitch.org/13314 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/e0492d57/attachment-0001.html From david.ponzone at gmail.com Wed Jun 30 06:25:28 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Wed, 30 Jun 2010 15:25:28 +0200 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> Message-ID: <7D2C6429-EB5F-4720-90EE-6B0C0410A34A@gmail.com> This looks like you have disable-transcoding set to true. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 30/06/2010 ? 15:07, Mark Campbell-Smith a ?crit : > Updating the configuration did not help. > > I'm not telling FS to use any other codecs other than what I have > specified below, so I'm not sure what I have done wrong. Below is the > full trace: > > > ------------------------------------------------------------------------ > INVITE sip:1020 at 192.168.1.120 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae > From: 1000 ;tag=2c7a518d12f9370eo0 > To: > Call-ID: 316156b9-f6b413d2 at 192.168.1.121 > CSeq: 102 INVITE > Max-Forwards: 70 > Proxy-Authorization: Digest > username="1000",realm="192.168.1.120",nonce="77836a7a-8447-11df-93a6- > d9ad5b204ca2",uri="sip: > 1020 > @192.168.1.120 > ",algorithm > = > MD5 > ,response > = > "c4a5e5cb60676366b8bcfb1329f3fc08 > ",qop=auth,nc=00000001,cnonce="6a0220d0" > Contact: 1000 > Expires: 240 > User-Agent: Linksys/PAP2T-5.1.6(LS) > Content-Length: 451 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura, replaces > Content-Type: application/sdp > > v=0 > o=- 17789112 17789112 IN IP4 192.168.1.121 > s=- > c=IN IP4 192.168.1.121 > t=0 0 > m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:102 G726-32/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:96 G726-40/8000 > a=rtpmap:97 G726-24/8000 > a=rtpmap:98 G726-16/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:30 > a=sendrecv > > ------------------------------------------------------------------------ > send 343 bytes to udp/[192.168.1.121]:5060 at 13:00:32.882684: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae > From: 1000 ;tag=2c7a518d12f9370eo0 > To: > Call-ID: 316156b9-f6b413d2 at 192.168.1.121 > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19 > 14-49-15 -0500 > Content-Length: 0 > > > ------------------------------------------------------------------------ > 2010-06-30 23:00:32.899423 [DEBUG] sofia.c:5975 IP 192.168.1.121 > Rejected by acl "domains". Falling back to Digest auth. > 2010-06-30 23:00:32.914917 [NOTICE] switch_channel.c:776 New Channel > sofia/internal/1000 at 192.168.1.120 > [77a3ab82-8447-11df-93a7-d9ad5b204ca2] > 2010-06-30 23:00:32.943135 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.1.120) Running State Change CS_NEW > 2010-06-30 23:00:32.945232 [DEBUG] switch_core_state_machine.c:320 > (sofia/internal/1000 at 192.168.1.120) State NEW > 2010-06-30 23:00:32.989778 [DEBUG] sofia.c:4293 Channel > sofia/internal/1000 at 192.168.1.120 entering state [received][100] > 2010-06-30 23:00:32.997500 [DEBUG] sofia.c:4304 Remote SDP: > v=0 > o=- 17789112 17789112 IN IP4 192.168.1.121 > s=- > c=IN IP4 192.168.1.121 > t=0 0 > m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:102 G726-32/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:96 G726-40/8000 > a=rtpmap:97 G726-24/8000 > a=rtpmap:98 G726-16/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:30 > > 2010-06-30 23:00:33.002369 [DEBUG] sofia_glue.c:3877 Audio Codec > Compare [PCMU:0:8000:30]/[G729:18:8000:20] > 2010-06-30 23:00:33.005967 [DEBUG] sofia_glue.c:3877 Audio Codec > Compare [PCMU:0:8000:30]/[PCMU:0:8000:20] > 2010-06-30 23:00:33.008248 [DEBUG] sofia_glue.c:3877 Audio Codec > Compare [PCMU:0:8000:30]/[GSM:3:8000:20] > 2010-06-30 23:00:33.010721 [DEBUG] sofia_glue.c:3924 Substituting > codec PCMU at 30i@8000h > 2010-06-30 23:00:33.019002 [DEBUG] sofia_glue.c:2462 Set Codec > sofia/internal/1000 at 192.168.1.120 PCMU/8000 30 ms 240 samples > 2010-06-30 23:00:33.038714 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf > send/recv payload to 101 > 2010-06-30 23:00:33.040992 [DEBUG] sofia.c:4451 > (sofia/internal/1000 at 192.168.1.120) State Change CS_NEW -> CS_INIT > 2010-06-30 23:00:33.043182 [DEBUG] switch_core_session.c:1027 Send > signal sofia/internal/1000 at 192.168.1.120 [BREAK] > 2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.1.120) Running State Change CS_INIT > 2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/1000 at 192.168.1.120) State INIT > 2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:83 > sofia/internal/1000 at 192.168.1.120 SOFIA INIT > 2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:117 > (sofia/internal/1000 at 192.168.1.120) State Change CS_INIT -> CS_ROUTING > 2010-06-30 23:00:33.048752 [DEBUG] switch_core_session.c:1027 Send > signal sofia/internal/1000 at 192.168.1.120 [BREAK] > 2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/1000 at 192.168.1.120) State INIT going to sleep > 2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.1.120) Running State Change CS_ROUTING > 2010-06-30 23:00:33.048752 [DEBUG] switch_channel.c:1474 > (sofia/internal/1000 at 192.168.1.120) Callstate Change DOWN -> RINGING > 2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/1000 at 192.168.1.120) State ROUTING > 2010-06-30 23:00:33.055456 [DEBUG] switch_channel.c:1333 > (sofia/internal/1000 at 192.168.1.120) Callstate Change RINGING -> ACTIVE > 2010-06-30 23:00:33.055456 [DEBUG] mod_sofia.c:140 > sofia/internal/1000 at 192.168.1.120 SOFIA ROUTING > 2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:77 > sofia/internal/1000 at 192.168.1.120 Standard ROUTING > 2010-06-30 23:00:33.055456 [INFO] mod_dialplan_xml.c:331 Processing > 1000->1020 in context default > Dialplan: sofia/internal/1000 at 192.168.1.120 parsing > [default->Local_1000_1019] continue=false > Dialplan: sofia/internal/1000 at 192.168.1.120 Regex (FAIL) > [Local_1000_1019] destination_number(1020) =~ /^(10[01][0-9])$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.1.120 parsing > [default->Mobile_1020s] continue=false > Dialplan: sofia/internal/1000 at 192.168.1.120 Regex (PASS) > [Mobile_1020s] destination_number(1020) =~ /^(102[0-9])$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.1.120 Action > set(dialed_extension=1020) > Dialplan: sofia/internal/1000 at 192.168.1.120 Action > export(codec_string=GSM) > Dialplan: sofia/internal/1000 at 192.168.1.120 Action > set(codec_string=GSM) > Dialplan: sofia/internal/1000 at 192.168.1.120 Action > bridge(user/${dialed_extension}@${domain}) > Dialplan: sofia/internal/1000 at 192.168.1.120 Action set_user(1000@$ > {domain}) > Dialplan: sofia/internal/1000 at 192.168.1.120 Action answer() > Dialplan: sofia/internal/1000 at 192.168.1.120 Action sleep(1000) > Dialplan: sofia/internal/1000 at 192.168.1.120 Action > system(/usr/local/freeswitch/scripts/sms.pl ${smsaccount} > ${smspassword} ${smsnumber} 'You have one new voicemail from ${effec$ > 2010-06-30 23:00:33.110926 [DEBUG] switch_core_state_machine.c:119 > (sofia/internal/1000 at 192.168.1.120) State Change CS_ROUTING -> > CS_EXECUTE > 2010-06-30 23:00:33.114917 [DEBUG] switch_core_session.c:1027 Send > signal sofia/internal/1000 at 192.168.1.120 [BREAK] > 2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/1000 at 192.168.1.120) State ROUTING going to sleep > 2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.1.120) Running State Change CS_EXECUTE > 2010-06-30 23:00:33.128360 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/1000 at 192.168.1.120) State EXECUTE > 2010-06-30 23:00:33.129661 [DEBUG] mod_sofia.c:233 > sofia/internal/1000 at 192.168.1.120 SOFIA EXECUTE > 2010-06-30 23:00:33.134566 [DEBUG] switch_core_state_machine.c:157 > sofia/internal/1000 at 192.168.1.120 Standard EXECUTE > EXECUTE sofia/internal/1000 at 192.168.1.120 set(dialed_extension=1020) > 2010-06-30 23:00:33.157698 [DEBUG] mod_dptools.c:843 > sofia/internal/1000 at 192.168.1.120 SET [dialed_extension]=[1020] > EXECUTE sofia/internal/1000 at 192.168.1.120 bridge(user/1020 at mydns.dyndns.org > ) > 2010-06-30 23:00:33.248458 [DEBUG] switch_ivr_originate.c:1956 > variable string 0 = [presence_id=1020 at mydns.dyndns.org] > 2010-06-30 23:00:33.260573 [NOTICE] switch_channel.c:776 New Channel > sofia/internal/sip:1020 at 192.168.1.123:5060 > [77d7e3b6-8447-11df-93a8-d9ad5b204ca2] > 2010-06-30 23:00:33.303487 [DEBUG] mod_sofia.c:3883 > (sofia/internal/sip:1020 at 192.168.1.123:5060) State Change CS_NEW -> > CS_INIT > 2010-06-30 23:00:33.304564 [DEBUG] switch_core_session.c:1027 Send > signal sofia/internal/sip:1020 at 192.168.1.123:5060 [BREAK] > 2010-06-30 23:00:33.327507 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/sip:1020 at 192.168.1.123:5060) Running State Change > CS_INIT > 2010-06-30 23:00:33.332010 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/sip:1020 at 192.168.1.123:5060) State INIT > 2010-06-30 23:00:33.334621 [DEBUG] mod_sofia.c:83 > sofia/internal/sip:1020 at 192.168.1.123:5060 SOFIA INIT > send 1183 bytes to udp/[192.168.1.123]:5060 at 13:00:33.351944: > > ------------------------------------------------------------------------ > INVITE sip:1020 at 192.168.1.123:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bKFDpKe5v6j2QcQ > Max-Forwards: 69 > From: "1000" ;tag=Z51Qve55SUHta > To: > Call-ID: 4f442091-feea-122d-448a-00e04c0312e9 > CSeq: 132833080 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19 > 14-49-15 -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 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 246 > X-FS-Support: update_display > Remote-Party-ID: "1000" > ;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1277875447 1277875448 IN IP4 192.168.1.120 > s=FreeSWITCH > c=IN IP4 192.168.1.120 > t=0 0 > m=audio 27386 RTP/AVP 0 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:30 > > It fails after this with Not Acceptable Here / > INCOMPATIBLE_DESTINATION > > On Mon, Jun 28, 2010 at 5:15 PM, David Ponzone > wrote: >> Please, retry with a genuine config (the default one would be a wise >> choice). >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou >> diffusion >> non autoris?e est interdite. Tout message ?lectronique est >> susceptible >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce >> message s'il >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire >> de ce >> message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur. >> >> >> >> Le 28/06/2010 ? 07:37, Mark Campbell-Smith a ?crit : >> >> Hi All, >> >> I'm not really sure if I got a firm answer for this one. Is the only >> way to ensure that transcoding is performed is by using >> late-negotiation? Why aren't all my codecs sent in the INVITE >> message >> to the B-leg (extension 1020)? >> >> Thanks! >> >> On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith >> wrote: >> >> FS version and codecs are shown below, but my config file are >> probably >> >> quite old. But I guess they should still work? >> >> All codecs are loaded, and the call works if late negotiation is set >> >> on profile internal. >> >> As I wrote above: >> >> The call setup is extension 1000 calls extension 1020 >> >> 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by >> >> FS as the A-leg codec >> >> 2. Extension 1020 only supports GSM codec. The call fails with Not >> >> Acceptable Here. >> >> I forgot to write that Extension 1000 does not support GSM (I want to >> >> force transcoding). Is that why FS is filtering out GSM on the b- >> leg? >> >> freeswitch at internal> version >> >> FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500) >> >> freeswitch at internal> show codecs >> >> type,name,ikey >> >> codec,ADPCM (IMA),mod_voipcodecs >> >> codec,G.711 alaw,CORE_PCM_MODULE >> >> codec,G.711 ulaw,CORE_PCM_MODULE >> >> codec,G.722,mod_voipcodecs >> >> codec,G.723.1 6.3k,mod_g723_1 >> >> codec,G.726 16k,mod_voipcodecs >> >> codec,G.726 16k (AAL2),mod_voipcodecs >> >> codec,G.726 24k,mod_voipcodecs >> >> codec,G.726 24k (AAL2),mod_voipcodecs >> >> codec,G.726 32k,mod_voipcodecs >> >> codec,G.726 32k (AAL2),mod_voipcodecs >> >> codec,G.726 40k,mod_voipcodecs >> >> codec,G.726 40k (AAL2),mod_voipcodecs >> >> codec,G.729,mod_com_g729 >> >> codec,GSM,mod_voipcodecs >> >> codec,H.261 Video (passthru),mod_h26x >> >> codec,H.263 Video (passthru),mod_h26x >> >> codec,H.263+ Video (passthru),mod_h26x >> >> codec,H.263++ Video (passthru),mod_h26x >> >> codec,H.264 Video (passthru),mod_h26x >> >> codec,LPC-10,mod_voipcodecs >> >> 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,Speex,mod_speex >> >> 25 total. >> >> >> On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone > > >> wrote: >> >> Mark, >> >> I confirm that, as I wrote that wiki page (the early negotiation >> part) :) >> >> Can you really confirm your FS version ? >> >> The parameter you showed is old. >> >> codec-prefs has been replaced in SIP profiles by: >> >> >> >> >> >> David Ponzone Direction Technique >> >> email: david.ponzone at ipeva.fr >> >> tel: 01 74 03 18 97 >> >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> >> tel: 0811 46 26 26 >> >> www.ipeva.fr - www.ipeva-studio.com >> >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis ? >> >> l'intention exclusive de ses destinataires. Toute utilisation ou >> diffusion >> >> non autoris?e est interdite. Tout message ?lectronique est >> susceptible >> >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce >> message s'il >> >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire >> de ce >> >> message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur. >> >> >> >> Le 23/06/2010 ? 14:43, Mark Campbell-Smith a ?crit : >> >> Check this good wiki page for how FS negotiates codecs (early >> >> negotiation default): >> >> http://wiki.freeswitch.org/wiki/Codec_Negotiation >> >> I have this set in my internal profile: >> >> >> >> and as stated before vars.xml: >> >> >> >> >> >> Setting late negotiation works (thanks Sergey), but reading the wiki >> >> page, I see the following sentence, which I interpret that GSM should >> >> still be sent: >> >> When FS calls leg B, the list of codecs in outbound-codec-prefs for >> >> the SIP profile is reorganized by pushing the codec negotiated above >> >> for leg A at the top . If B does not accept any of the codecs, the >> >> calls fails, obviously. >> >> >> >> On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano >> >> wrote: >> >> I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? >> >> Wouldn't fs only get involved when its media server is referred to? >> >> If the "other endpoint" will only accept G729, doesn't that mean you >> >> need to change that endpoint to also accept G711 or also license G729 >> >> in FS? >> >> On 6/23/10, Mark Campbell-Smith wrote: >> >> Test Setup: >> >> vars.xml: >> >> >> >> > data="outbound_codec_prefs=G729,PCMU,GSM"/> >> >> The call setup is extension 1000 calls extension 1020 >> >> 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by >> >> FS as the A-leg codec >> >> 2. Extension 1020 only supports GSM codec. The call fails with Not >> >> Acceptable Here. >> >> FS only offers G729 and PCMU to 1020. How do I change the number of >> >> codecs that are offered to an extension? I know I can change the >> >> order in the codec_prefs, but would prefer FS to offer all three >> >> codecs to an extension. >> >> m=audio 23662 RTP/AVP 0 18 101 13 >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:18 G729/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=rtpmap:13 CN/8000 >> >> a=ptime:30 >> >> Thanks >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> -- >> >> Sent from my mobile device >> >> ====================== >> >> Tony Graziano, Manager >> >> Telephone: 434.984.8430 >> >> sip: tgraziano at voice.myitdepartment.net >> >> Fax: 434.984.8431 >> >> Email: tgraziano at myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> >> Telephone: 434.984.8426 >> >> sip: helpdesk at voice.myitdepartment.net >> >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> >> http://www.myitdepartment.net/gethelp/ >> >> Why do mathematicians always confuse Halloween and Christmas? >> >> Because 31 Oct = 25 Dec. >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.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 >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.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 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100630/e1e50e29/attachment-0001.html From testeador01 at gmail.com Wed Jun 30 06:35:44 2010 From: testeador01 at gmail.com (Milena) Date: Wed, 30 Jun 2010 08:35:44 -0500 Subject: [Freeswitch-users] Connect two FreeSWITCH Boxes In-Reply-To: <053901cb184a$02596290$070c27b0$@com> References: <053901cb184a$02596290$070c27b0$@com> Message-ID: Hello Specifically, check the dialplan for this extension: "Calls from BoxB" or post it so we can look at it and help you, but please don't do it here, use pastebin!!: pastebin.freeswitch.org then send us the link. -Milena 2010/6/30 Peder > I would guess there is an error in public on A, but you didn?t include > all of that context so I can?t say for sure. Note that it matches 10001 in > public, but when it gets to Local_Call, it just sees $1, not 10001. That s > why the match is failing. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/9ec9dfe1/attachment.html From anthony.minessale at gmail.com Wed Jun 30 06:38:28 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Jun 2010 08:38:28 -0500 Subject: [Freeswitch-users] Possible Mem LeakFS In-Reply-To: References: Message-ID: You sure about that? There are more bytes used in this email thread than that valgrind report shows lost? We have many people using FreeSWITCH in high density scenarios and nobody else is complaining. FreeSWITCH uses some memory pooling so its natural for it to retain some memory in certain cases. On Wed, Jun 30, 2010 at 8:07 AM, Deya M wrote: > Hi, > > Starting with total memory usage of 300k, I go up to 2GB after 10 hours > running FS. > > Just doing further tests, I got the log file, which at the end has: > > > ==4573== LEAK SUMMARY: > ==4573== definitely lost: 8,166 bytes in 26 blocks > ==4573== indirectly lost: 284 bytes in 1 blocks > ==4573== possibly lost: 2,794,680 bytes in 477 blocks > ==4573== still reachable: 74,666 bytes in 2,057 blocks > ==4573== suppressed: 0 bytes in 0 blocks > ==4573== > ==4573== ERROR SUMMARY: 232006 errors from 122 contexts (suppressed: 326 > from 13) > > Appreciate your help! > > > > http://pastebin.freeswitch.org/13314 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100630/0d0d753f/attachment.html From mcampbellsmith at gmail.com Wed Jun 30 06:46:23 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 30 Jun 2010 23:46:23 +1000 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: <7D2C6429-EB5F-4720-90EE-6B0C0410A34A@gmail.com> References: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> <7D2C6429-EB5F-4720-90EE-6B0C0410A34A@gmail.com> Message-ID: <14709fd5-c2ce-4627-abd9-e33bcc8b0d03@email.android.com> Nope. I have transcoding enabled. At least I have that line disabing transcoding commented out in internal.xml and I assume by default FS enables transcoding -- Sent from my Android phone with K-9 Mail. Please excuse my brevity. From testeador01 at gmail.com Wed Jun 30 06:48:18 2010 From: testeador01 at gmail.com (Milena) Date: Wed, 30 Jun 2010 08:48:18 -0500 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 In-Reply-To: References: Message-ID: PASTEBIN!!! 2010/6/29 Ravi Kuru > > this is the freeswitch.log: > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/3abf52de/attachment.html From mkane02 at harris.com Wed Jun 30 06:49:02 2010 From: mkane02 at harris.com (Kane, Michael (mkane02)) Date: Wed, 30 Jun 2010 09:49:02 -0400 Subject: [Freeswitch-users] Connect two FreeSWITCH Boxes In-Reply-To: References: <053901cb184a$02596290$070c27b0$@com> Message-ID: Sorry about that, I figured out the process of pasting after the fact. I found the problem, as usual it was due to that thing called a human and his fingers not coordinating with his brain. Again sorry for posting my dirty laundry here on the list. Mike From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena Sent: Wednesday, June 30, 2010 9:36 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Connect two FreeSWITCH Boxes Hello Specifically, check the dialplan for this extension: "Calls from BoxB" or post it so we can look at it and help you, but please don't do it here, use pastebin!!: pastebin.freeswitch.org then send us the link. -Milena 2010/6/30 Peder I would guess there is an error in public on A, but you didn't include all of that context so I can't say for sure. Note that it matches 10001 in public, but when it gets to Local_Call, it just sees $1, not 10001. That s why the match is failing. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/d5178e57/attachment.html From brian at freeswitch.org Wed Jun 30 06:52:56 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Jun 2010 08:52:56 -0500 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 In-Reply-To: References: Message-ID: Please do not reply to digest emails. Subscribe normal please. Its hard to follow you if you do this. /b On Jun 29, 2010, at 11:46 AM, Ravi Kuru wrote: > Hi Steve, > > I put install the jdk 64 bit now, i don't get that error but PhoneTest.java does not working > > i compile PhoneTest.java without error and freeswitch.log i see that trying to PhoneTest but it did not work. > do you know why it is not working? From david.ponzone at gmail.com Wed Jun 30 06:53:59 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Wed, 30 Jun 2010 15:53:59 +0200 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: <14709fd5-c2ce-4627-abd9-e33bcc8b0d03@email.android.com> References: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> <7D2C6429-EB5F-4720-90EE-6B0C0410A34A@gmail.com> <14709fd5-c2ce-4627-abd9-e33bcc8b0d03@email.android.com> Message-ID: can you send us the output of: sofia status profile internal (of course, hide IPs and else if required). David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 30/06/2010 ? 15:46, Mark Campbell-Smith a ?crit : > Nope. I have transcoding enabled. At least I have that line disabing > transcoding commented out in internal.xml and I assume by default FS > enables transcoding > > > -- > Sent from my Android phone with K-9 Mail. Please excuse my brevity. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/ba815ff2/attachment-0001.html From sameer2k3t at gmail.com Wed Jun 30 07:29:53 2010 From: sameer2k3t at gmail.com (Sameer Khan) Date: Wed, 30 Jun 2010 19:29:53 +0500 Subject: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen In-Reply-To: References: Message-ID: Somebody reply please On Tue, Jun 29, 2010 at 9:11 PM, Sameer Khan wrote: > Hi > Can mod_dingaling be used for multiple outgoing calls to google network > like skypopen.? If yes please reply > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/f8a64bc2/attachment.html From nagalenoj at gmail.com Wed Jun 30 07:37:14 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Wed, 30 Jun 2010 20:07:14 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: Dear Anthony, I've tried using the group_confirm_cancel_timeout as per the discussion we had in IRC. You wanted to used it as part of dial string and not as a channel variable. But, It doesn't work for me. Here is how I've given the commands and the script I've executed. Even when I give group_confirm_cancel_timeout, the callee's leg is getting disconnected after legtimeout. connect sendmsg call-command: execute execute-app-name:answer sendmsg call-command: execute execute-app-name: set execute-app-arg: group_confirm_key=exec sendmsg call-command: execute execute-app-name: set execute-app-arg: group_confirm_file=perl /root/bridge.pl sendmsg call-command: execute execute-app-name: bridge execute-app-arg: {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 bridge.pl: #!/usr/bin/perl use freeswitch; our $session; freeswitch::consoleLog("info","Goint to get the digits"); # To simulate the scenario I used sleep here. sleep(30); 1; Kindly tell me whats wrong in the above. On Fri, Jun 18, 2010 at 7:44 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I said leg timeout beats the group confirm timeouts > > group_confirm_cancel_timeout is a whole different variable, when you set > that to true it will stop all the timeouts as soon as you reach > group_confirm execution > > {group_confirm_cancel_timeout=true}[leg_timeout=10]sofia/foo/foo at bar.com > > > > On Fri, Jun 18, 2010 at 12:50 AM, lakshmanan ganapathy < > lakindia89 at gmail.com> wrote: > >> Dear Antony, >> >> Also in the leg_timeout wiki >> http://wiki.freeswitch.org/wiki/Variable_leg_timeout, it is stated as >> follows >> >> "If you are using group confirm then you can cancel the timeout by using >> the group_confirm_cancel_timeoutchannel variable." >> >> >> >> On Thu, Jun 17, 2010 at 8:22 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> no there is no way, besides making both timeouts longer. >>> you could file a feature request/bounty to ask for a feature to stop the >>> leg timer when you reach the confirm. >>> >>> >>> On Thu, Jun 17, 2010 at 4:23 AM, Nagalenoj H. wrote: >>> >>>> Anthony, >>>> But, then there is no use. Am I right? Usually, we'll use the >>>> group_confirm_cancel_timeout only when we need to override the leg_timeout. >>>> But it happens in reverse in this case., >>>> >>>> I've tried using the group_confirm_cancel_timeout along with >>>> call_timeout and things happening similar like setting leg_timout. >>>> >>>> Then, tried without setting leg_timeout and call_timeout explicitly. >>>> * In this case if the callee doesn't picks the call, it >>>> disconnects the leg in 30 secs. >>>> * If he answers the call and the script continues to execute, >>>> the leg is disconnected in 60 secs. >>>> >>>> What I need to do is, when the callee picks the call the leg_timeout >>>> should not be accounted more and the leg shouldn't be disconnected because >>>> of leg_timeout after that. >>>> >>>> Any other way of doing this?! >>>> >>>> >>>> >>>> On Tue, Jun 15, 2010 at 10:53 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> leg timeout beats the group confirm timeouts >>>>> >>>>> >>>>> On Tue, Jun 15, 2010 at 12:28 AM, Nagalenoj H. wrote: >>>>> >>>>>> Dear friends, >>>>>> I've tried using the group_confirm_cancel_timeout channel >>>>>> variable. I've written a testing script to get digits before bridging. But, >>>>>> it doesn't seem to be working. >>>>>> >>>>>> My understanding after reading wiki is, >>>>>> * When I dial [leg_timeout=10]user/1005, if he answers before >>>>>> timeout and in the process of giving digits, then the call shouldn't be >>>>>> disconnected after the leg_timeout secs (10 sec in the example). >>>>>> >>>>>> But, When I experiment it, the call is getting disconnected after 10 >>>>>> seconds and it doesn't bother whether the callee has answered the >>>>>> call(Started giving digits) or not answered at all. >>>>>> >>>>>> I've checked it with nc as follows, >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_key=exec >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_file=perl /root/confirm.pl >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_cancel_timeout=1 >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: bridge >>>>>> execute-app-arg: [leg_timeout=10]user/1005 >>>>>> >>>>>> And here is the script, >>>>>> >>>>>> use freeswitch; >>>>>> our $session; >>>>>> my $digit; >>>>>> >>>>>> while(1) { >>>>>> # Wait till response timeout for the first digit. >>>>>> $digit = $session->getDigits(1, "", 10000); >>>>>> freeswitch::consoleLog ("info","Digit>>".$digit."<<"); >>>>>> >>>>>> if (! $session->ready() ) { >>>>>> freeswitch::consoleLog("info","Going to Exit\n"); >>>>>> last; >>>>>> } >>>>>> if (defined $digit and $digit ne "" ) { >>>>>> freeswitch::consoleLog("info","DTMF received: >>>>>> $digit\n"); >>>>>> if ($digit eq '#') { >>>>>> return; >>>>>> } >>>>>> } >>>>>> else { >>>>>> freeswitch::consoleLog("info","Timeout\n"); >>>>>> $session->hangup(); >>>>>> } >>>>>> } >>>>>> 1; >>>>>> >>>>>> If my understanding is right then, I believe there is something wrong >>>>>> with channel_variable. >>>>>> >>>>>> Kindly help me to resolve this. >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> Nagalenoj H. >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> Nagalenoj H. >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > > -- > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/b1f12b46/attachment-0001.html From brian at freeswitch.org Wed Jun 30 07:38:03 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Jun 2010 09:38:03 -0500 Subject: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen In-Reply-To: References: Message-ID: <35A36D3E-7B26-455B-B398-6ABCBFA1515E@freeswitch.org> Have you tried it? its also best to let someone have a chance to reply and not get impatient on the list. Someone will answer you if they care enough about your issue. While you can do this with mod_dingaling you'll get smacked about by Google if you try doing too many at once.. or too many in a short period of time. /b On Jun 30, 2010, at 9:29 AM, Sameer Khan wrote: > Somebody reply please From freeswitch-users at digitaldan.com Wed Jun 30 07:39:27 2010 From: freeswitch-users at digitaldan.com (Dan) Date: Wed, 30 Jun 2010 08:39:27 -0600 (MDT) Subject: [Freeswitch-users] calls ending with MEDIA_TIMEOUT In-Reply-To: Message-ID: <23029141.3084.1277908748262.JavaMail.daniel@radio> Thanks for your response, I put everything up on pastebin http://pastebin.freeswitch.org/13322 . The application in question is actually javascript, I'm using lua in production but was switching to the posted js version with the upgrade. Now that I posted it i realized I have in my dial plan, I believed I used it in the older version to get around some dtmf issues or choppy playback (can't remember), not sure if this could be part of the issue (although it works fine in the production version I'm running) So I pulled one of the recordings that hung up after 4 minutes, but was only 24 seconds long, it sounded fine (but obviously too short). But another one that dropped after 5 minutes and only 19 seconds in length was very choppy and included short spurts of audio from parts of the call that were much longer then 19 seconds. From: "Anthony Minessale" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 29, 2010 12:54:21 PM Subject: Re: [Freeswitch-users] calls ending with MEDIA_TIMEOUT it's not 100% accurate in the media timeout. It would be too expensive to use actual timers, it uses the number of samples you should be getting from rtp and a number of loops where no media was received. Migrating from svn 13000 range to GIT is a big step and you may have to adjust to some new behaviors. media_timeout may not even have existed that long ago I don't recall. If you don't need media timeouts turn off the param or turn it up to longer. On Tue, Jun 29, 2010 at 1:09 PM, Michael Collins < msc at freeswitch.org > wrote: Pastebin your dialplan and the lua script for starters. Also, is it the 5300 that is responding with the media timeout? -MC On Tue, Jun 29, 2010 at 10:15 AM, Dan < freeswitch-users at digitaldan.com > wrote: Hi guys, I have been running two freeswitch boxes (13754M) that answer calls from a cisco 5300 (both on the same network) and records them to disk with a small lua application. This has been working well for the past few months. I decided to upgrade one of them to trunk ( git-3fbd9e2 2010-06-11 11-08-51 -0500 ) and have run into a problem. Some calls will fail with a MEDIA_TIMEOUT after a few minutes, the time it takes to fail ranges from 4 minutes to 10 minutes, I don't have a full sip trace or pcap dump yet, I reverted back to the old freeswitch version (on the same hardware) and have not been able to reproduce it in a test environment yet ( I continue to try). Below are the relevant lines from the log files for one of the calls: 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2257 (sofia/external/ nobody at 192.168.21.4 ) Callstate Change ACTIVE -> HANGUP 2010-06-23 07:42:19.033466 [NOTICE] mod_sofia.c:884 Hangup sofia/external/ nobody at 192.168.21.4 [CS_EXECUTE] [MEDIA_TIMEOUT] 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2273 Send signal sofia/external/ nobody at 192.168.21.4 [KILL] 2010-06-23 07:42:19.033466 [DEBUG] switch_core_session.c:1023 Send signal sofia/external/ nobody at 192.168.21.4 [BREAK] 2010-06-23 07:42:19.033466 [DEBUG] switch_core_codec.c:146 sofia/external/ nobody at 192.168.21.4 Restore previous codec PCMU:0. My configuration is bone stock, so the rtp timeout value is at 300, but I have some calls that have lasted only 4 minutes. One other piece of information is that on one of the recordings that was hung up after 4 minutes and 17 seconds the recorded file was only 24 seconds long (it stopped recording after the first 24 seconds) , so I'm assuming freeswitch did not think there were any rtp packets to record. Any ideas on where to start debugging this? I have setup a new freeswitch box connected to the same 5300 to reproduce, but have not been able to generate the call volume ( there where around 30 calls being recorded) yet, but I'm working on it. Thanks! _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- 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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/ccb100ac/attachment.html From gmaruzz at celliax.org Wed Jun 30 07:40:58 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 30 Jun 2010 16:40:58 +0200 Subject: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen In-Reply-To: References: Message-ID: reply On Wed, Jun 30, 2010 at 4:29 PM, Sameer Khan wrote: > Somebody reply please > > On Tue, Jun 29, 2010 at 9:11 PM, Sameer Khan wrote: >> >> Hi >> Can mod_dingaling be used for multiple outgoing calls to google network >> like skypopen.? If yes please reply > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From testeador01 at gmail.com Wed Jun 30 07:42:00 2010 From: testeador01 at gmail.com (Milena) Date: Wed, 30 Jun 2010 09:42:00 -0500 Subject: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen In-Reply-To: References: Message-ID: If you're in a rush, you should test it yourself, if not then have some patience and wait for someone who knows to come in and reply.... 2010/6/30 Sameer Khan > Somebody reply please > > > On Tue, Jun 29, 2010 at 9:11 PM, Sameer Khan wrote: > >> Hi >> Can mod_dingaling be used for multiple outgoing calls to google network >> like skypopen.? If yes please reply >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/e334624a/attachment.html From deya787 at gmail.com Wed Jun 30 07:47:07 2010 From: deya787 at gmail.com (Deya M) Date: Wed, 30 Jun 2010 17:47:07 +0300 Subject: [Freeswitch-users] Possible Mem LeakFS Message-ID: Hi Anthony, I am starting with 300mb memory used, leaving the system before running fs for 2 hours, memory is not increasing. As soon as I start FS, it starts going up gradually, until 2GB, and still increasing. Is the a standard expected behaviour for allocating pools ? Bear in mind, that I am not using FS, connecting or doing any activities with the machine. The machine is almost 100 IDLE. Thanks, D:- 2010/6/30 > 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: Connect two FreeSWITCH Boxes (Milena) > 2. Re: Possible Mem LeakFS (Anthony Minessale) > 3. Re: Number of codecs offerred in SDP (Mark Campbell-Smith) > 4. Re: FreeSWITCH-users Digest, Vol 48, Issue 194 (Milena) > 5. Re: Connect two FreeSWITCH Boxes (Kane, Michael (mkane02)) > 6. Re: FreeSWITCH-users Digest, Vol 48, Issue 194 (Brian West) > 7. Re: Number of codecs offerred in SDP (David Ponzone) > > > ---------- Forwarded message ---------- > From: Milena > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 08:35:44 -0500 > Subject: Re: [Freeswitch-users] Connect two FreeSWITCH Boxes > Hello > > Specifically, check the dialplan for this extension: "Calls from BoxB" or > post it so we can look at it and help you, but please don't do it here, use > pastebin!!: pastebin.freeswitch.org then send us the link. > > -Milena > > 2010/6/30 Peder > >> I would guess there is an error in public on A, but you didn?t include >> all of that context so I can?t say for sure. Note that it matches 10001 in >> public, but when it gets to Local_Call, it just sees $1, not 10001. That s >> why the match is failing. >> > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 08:38:28 -0500 > Subject: Re: [Freeswitch-users] Possible Mem LeakFS > You sure about that? > There are more bytes used in this email thread than that valgrind report > shows lost? > We have many people using FreeSWITCH in high density scenarios and nobody > else is complaining. > FreeSWITCH uses some memory pooling so its natural for it to retain some > memory in certain cases. > > > On Wed, Jun 30, 2010 at 8:07 AM, Deya M wrote: > >> Hi, >> >> Starting with total memory usage of 300k, I go up to 2GB after 10 hours >> running FS. >> >> Just doing further tests, I got the log file, which at the end has: >> >> >> ==4573== LEAK SUMMARY: >> ==4573== definitely lost: 8,166 bytes in 26 blocks >> ==4573== indirectly lost: 284 bytes in 1 blocks >> ==4573== possibly lost: 2,794,680 bytes in 477 blocks >> ==4573== still reachable: 74,666 bytes in 2,057 blocks >> ==4573== suppressed: 0 bytes in 0 blocks >> ==4573== >> ==4573== ERROR SUMMARY: 232006 errors from 122 contexts (suppressed: 326 >> from 13) >> >> Appreciate your help! >> >> >> >> http://pastebin.freeswitch.org/13314 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > > ---------- Forwarded message ---------- > From: Mark Campbell-Smith > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 23:46:23 +1000 > Subject: Re: [Freeswitch-users] Number of codecs offerred in SDP > Nope. I have transcoding enabled. At least I have that line disabing > transcoding commented out in internal.xml and I assume by default FS enables > transcoding > > > -- > Sent from my Android phone with K-9 Mail. Please excuse my brevity. > > > > > ---------- Forwarded message ---------- > From: Milena > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 08:48:18 -0500 > Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 > > PASTEBIN!!! > > > 2010/6/29 Ravi Kuru > >> >> this is the freeswitch.log: >> > > > ---------- Forwarded message ---------- > From: "Kane, Michael (mkane02)" > To: > Date: Wed, 30 Jun 2010 09:49:02 -0400 > Subject: Re: [Freeswitch-users] Connect two FreeSWITCH Boxes > > Sorry about that, I figured out the process of pasting after the fact. I > found the problem, as usual it was due to that thing called a human and his > fingers not coordinating with his brain. Again sorry for posting my dirty > laundry here on the list. > > > > Mike > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Milena > *Sent:* Wednesday, June 30, 2010 9:36 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Connect two FreeSWITCH Boxes > > > > Hello > > > > Specifically, check the dialplan for this extension: "Calls from BoxB" or > post it so we can look at it and help you, but please don't do it here, use > pastebin!!: pastebin.freeswitch.org then send us the link. > > > > -Milena > > > > 2010/6/30 Peder > > I would guess there is an error in public on A, but you didn?t include all > of that context so I can?t say for sure. Note that it matches 10001 in > public, but when it gets to Local_Call, it just sees $1, not 10001. That s > why the match is failing. > > > ---------- Forwarded message ---------- > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 08:52:56 -0500 > Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 > Please do not reply to digest emails. Subscribe normal please. Its hard > to follow you if you do this. > > /b > > On Jun 29, 2010, at 11:46 AM, Ravi Kuru wrote: > > > Hi Steve, > > > > I put install the jdk 64 bit now, i don't get that error but > PhoneTest.java does not working > > > > i compile PhoneTest.java without error and freeswitch.log i see that > trying to PhoneTest but it did not work. > > do you know why it is not working? > > > > > > ---------- Forwarded message ---------- > From: David Ponzone > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 15:53:59 +0200 > Subject: Re: [Freeswitch-users] Number of codecs offerred in SDP > can you send us the output of: > sofia status profile internal > (of course, hide IPs and else if required). > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 30/06/2010 ? 15:46, Mark Campbell-Smith a ?crit : > > Nope. I have transcoding enabled. At least I have that line disabing > transcoding commented out in internal.xml and I assume by default FS enables > transcoding > > > -- > Sent from my Android phone with K-9 Mail. Please excuse my brevity. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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/20100630/f5d0018c/attachment-0001.html From anthony.minessale at gmail.com Wed Jun 30 08:00:41 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Jun 2010 10:00:41 -0500 Subject: [Freeswitch-users] Possible Mem LeakFS In-Reply-To: References: Message-ID: What OS and revision of FS are you using? You may want to build the latest revision of FreeSWITCH and try again. That certainly does not happen on the many installs we have of FreeSWITCH and I am sure our other users would report such a significant issue. On Wed, Jun 30, 2010 at 9:47 AM, Deya M wrote: > Hi Anthony, > > I am starting with 300mb memory used, leaving the system before running fs > for 2 hours, memory is not increasing. > As soon as I start FS, it starts going up gradually, until 2GB, and still > increasing. > > Is the a standard expected behaviour for allocating pools ? > > Bear in mind, that I am not using FS, connecting or doing any activities > with the machine. The machine is almost 100 IDLE. > > Thanks, > > D:- > > 2010/6/30 > >> 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: Connect two FreeSWITCH Boxes (Milena) >> 2. Re: Possible Mem LeakFS (Anthony Minessale) >> 3. Re: Number of codecs offerred in SDP (Mark Campbell-Smith) >> 4. Re: FreeSWITCH-users Digest, Vol 48, Issue 194 (Milena) >> 5. Re: Connect two FreeSWITCH Boxes (Kane, Michael (mkane02)) >> 6. Re: FreeSWITCH-users Digest, Vol 48, Issue 194 (Brian West) >> 7. Re: Number of codecs offerred in SDP (David Ponzone) >> >> >> ---------- Forwarded message ---------- >> From: Milena >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 30 Jun 2010 08:35:44 -0500 >> Subject: Re: [Freeswitch-users] Connect two FreeSWITCH Boxes >> Hello >> >> Specifically, check the dialplan for this extension: "Calls from BoxB" or >> post it so we can look at it and help you, but please don't do it here, use >> pastebin!!: pastebin.freeswitch.org then send us the link. >> >> -Milena >> >> 2010/6/30 Peder >> >>> I would guess there is an error in public on A, but you didn?t include >>> all of that context so I can?t say for sure. Note that it matches 10001 in >>> public, but when it gets to Local_Call, it just sees $1, not 10001. That s >>> why the match is failing. >>> >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 30 Jun 2010 08:38:28 -0500 >> Subject: Re: [Freeswitch-users] Possible Mem LeakFS >> You sure about that? >> There are more bytes used in this email thread than that valgrind report >> shows lost? >> We have many people using FreeSWITCH in high density scenarios and nobody >> else is complaining. >> FreeSWITCH uses some memory pooling so its natural for it to retain some >> memory in certain cases. >> >> >> On Wed, Jun 30, 2010 at 8:07 AM, Deya M wrote: >> >>> Hi, >>> >>> Starting with total memory usage of 300k, I go up to 2GB after 10 hours >>> running FS. >>> >>> Just doing further tests, I got the log file, which at the end has: >>> >>> >>> ==4573== LEAK SUMMARY: >>> ==4573== definitely lost: 8,166 bytes in 26 blocks >>> ==4573== indirectly lost: 284 bytes in 1 blocks >>> ==4573== possibly lost: 2,794,680 bytes in 477 blocks >>> ==4573== still reachable: 74,666 bytes in 2,057 blocks >>> ==4573== suppressed: 0 bytes in 0 blocks >>> ==4573== >>> ==4573== ERROR SUMMARY: 232006 errors from 122 contexts (suppressed: 326 >>> from 13) >>> >>> Appreciate your help! >>> >>> >>> >>> http://pastebin.freeswitch.org/13314 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> >> ---------- Forwarded message ---------- >> From: Mark Campbell-Smith >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 30 Jun 2010 23:46:23 +1000 >> Subject: Re: [Freeswitch-users] Number of codecs offerred in SDP >> Nope. I have transcoding enabled. At least I have that line disabing >> transcoding commented out in internal.xml and I assume by default FS enables >> transcoding >> >> >> -- >> Sent from my Android phone with K-9 Mail. Please excuse my brevity. >> >> >> >> >> ---------- Forwarded message ---------- >> From: Milena >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 30 Jun 2010 08:48:18 -0500 >> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 >> >> PASTEBIN!!! >> >> >> 2010/6/29 Ravi Kuru >> >>> >>> this is the freeswitch.log: >>> >> >> >> ---------- Forwarded message ---------- >> From: "Kane, Michael (mkane02)" >> To: >> Date: Wed, 30 Jun 2010 09:49:02 -0400 >> Subject: Re: [Freeswitch-users] Connect two FreeSWITCH Boxes >> >> Sorry about that, I figured out the process of pasting after the fact. I >> found the problem, as usual it was due to that thing called a human and his >> fingers not coordinating with his brain. Again sorry for posting my dirty >> laundry here on the list. >> >> >> >> Mike >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Milena >> *Sent:* Wednesday, June 30, 2010 9:36 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Connect two FreeSWITCH Boxes >> >> >> >> Hello >> >> >> >> Specifically, check the dialplan for this extension: "Calls from BoxB" or >> post it so we can look at it and help you, but please don't do it here, use >> pastebin!!: pastebin.freeswitch.org then send us the link. >> >> >> >> -Milena >> >> >> >> 2010/6/30 Peder >> >> I would guess there is an error in public on A, but you didn?t include all >> of that context so I can?t say for sure. Note that it matches 10001 in >> public, but when it gets to Local_Call, it just sees $1, not 10001. That s >> why the match is failing. >> >> >> ---------- Forwarded message ---------- >> From: Brian West >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 30 Jun 2010 08:52:56 -0500 >> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 >> Please do not reply to digest emails. Subscribe normal please. Its hard >> to follow you if you do this. >> >> /b >> >> On Jun 29, 2010, at 11:46 AM, Ravi Kuru wrote: >> >> > Hi Steve, >> > >> > I put install the jdk 64 bit now, i don't get that error but >> PhoneTest.java does not working >> > >> > i compile PhoneTest.java without error and freeswitch.log i see that >> trying to PhoneTest but it did not work. >> > do you know why it is not working? >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: David Ponzone >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 30 Jun 2010 15:53:59 +0200 >> Subject: Re: [Freeswitch-users] Number of codecs offerred in SDP >> can you send us the output of: >> sofia status profile internal >> (of course, hide IPs and else if required). >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur.* >> * >> * >> >> >> >> Le 30/06/2010 ? 15:46, Mark Campbell-Smith a ?crit : >> >> Nope. I have transcoding enabled. At least I have that line disabing >> transcoding commented out in internal.xml and I assume by default FS enables >> transcoding >> >> >> -- >> Sent from my Android phone with K-9 Mail. Please excuse my brevity. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100630/a07422a1/attachment-0001.html From jan.berger at video24.no Wed Jun 30 08:14:27 2010 From: jan.berger at video24.no (Jan Berger) Date: Wed, 30 Jun 2010 17:14:27 +0200 Subject: [Freeswitch-users] Possible Mem LeakFS In-Reply-To: References: Message-ID: <2E4194BF4AB24B9DBBA18B3FF8A4834C@dell9400> Deya, Stupid question ? you are not accidentally running a console app that printf into memory ? this sounds very much like a printf issue. Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Deya M Sent: 30. juni 2010 16:47 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Possible Mem LeakFS Hi Anthony, I am starting with 300mb memory used, leaving the system before running fs for 2 hours, memory is not increasing. As soon as I start FS, it starts going up gradually, until 2GB, and still increasing. Is the a standard expected behaviour for allocating pools ? Bear in mind, that I am not using FS, connecting or doing any activities with the machine. The machine is almost 100 IDLE. Thanks, D:- 2010/6/30 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: Connect two FreeSWITCH Boxes (Milena) 2. Re: Possible Mem LeakFS (Anthony Minessale) 3. Re: Number of codecs offerred in SDP (Mark Campbell-Smith) 4. Re: FreeSWITCH-users Digest, Vol 48, Issue 194 (Milena) 5. Re: Connect two FreeSWITCH Boxes (Kane, Michael (mkane02)) 6. Re: FreeSWITCH-users Digest, Vol 48, Issue 194 (Brian West) 7. Re: Number of codecs offerred in SDP (David Ponzone) ---------- Forwarded message ---------- From: Milena To: freeswitch-users at lists.freeswitch.org Date: Wed, 30 Jun 2010 08:35:44 -0500 Subject: Re: [Freeswitch-users] Connect two FreeSWITCH Boxes Hello Specifically, check the dialplan for this extension: "Calls from BoxB" or post it so we can look at it and help you, but please don't do it here, use pastebin!!: pastebin.freeswitch.org then send us the link. -Milena 2010/6/30 Peder I would guess there is an error in public on A, but you didn?t include all of that context so I can?t say for sure. Note that it matches 10001 in public, but when it gets to Local_Call, it just sees $1, not 10001. That s why the match is failing. ---------- Forwarded message ---------- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Date: Wed, 30 Jun 2010 08:38:28 -0500 Subject: Re: [Freeswitch-users] Possible Mem LeakFS You sure about that? There are more bytes used in this email thread than that valgrind report shows lost? We have many people using FreeSWITCH in high density scenarios and nobody else is complaining. FreeSWITCH uses some memory pooling so its natural for it to retain some memory in certain cases. On Wed, Jun 30, 2010 at 8:07 AM, Deya M wrote: Hi, Starting with total memory usage of 300k, I go up to 2GB after 10 hours running FS. Just doing further tests, I got the log file, which at the end has: ==4573== LEAK SUMMARY: ==4573== definitely lost: 8,166 bytes in 26 blocks ==4573== indirectly lost: 284 bytes in 1 blocks ==4573== possibly lost: 2,794,680 bytes in 477 blocks ==4573== still reachable: 74,666 bytes in 2,057 blocks ==4573== suppressed: 0 bytes in 0 blocks ==4573== ==4573== ERROR SUMMARY: 232006 errors from 122 contexts (suppressed: 326 from 13) Appreciate your help! http://pastebin.freeswitch.org/13314 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 ---------- Forwarded message ---------- From: Mark Campbell-Smith To: freeswitch-users at lists.freeswitch.org Date: Wed, 30 Jun 2010 23:46:23 +1000 Subject: Re: [Freeswitch-users] Number of codecs offerred in SDP Nope. I have transcoding enabled. At least I have that line disabing transcoding commented out in internal.xml and I assume by default FS enables transcoding -- Sent from my Android phone with K-9 Mail. Please excuse my brevity. ---------- Forwarded message ---------- From: Milena To: freeswitch-users at lists.freeswitch.org Date: Wed, 30 Jun 2010 08:48:18 -0500 Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 PASTEBIN!!! 2010/6/29 Ravi Kuru this is the freeswitch.log: ---------- Forwarded message ---------- From: "Kane, Michael (mkane02)" To: Date: Wed, 30 Jun 2010 09:49:02 -0400 Subject: Re: [Freeswitch-users] Connect two FreeSWITCH Boxes Sorry about that, I figured out the process of pasting after the fact. I found the problem, as usual it was due to that thing called a human and his fingers not coordinating with his brain. Again sorry for posting my dirty laundry here on the list. Mike From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena Sent: Wednesday, June 30, 2010 9:36 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Connect two FreeSWITCH Boxes Hello Specifically, check the dialplan for this extension: "Calls from BoxB" or post it so we can look at it and help you, but please don't do it here, use pastebin!!: pastebin.freeswitch.org then send us the link. -Milena 2010/6/30 Peder I would guess there is an error in public on A, but you didn?t include all of that context so I can?t say for sure. Note that it matches 10001 in public, but when it gets to Local_Call, it just sees $1, not 10001. That s why the match is failing. ---------- Forwarded message ---------- From: Brian West To: freeswitch-users at lists.freeswitch.org Date: Wed, 30 Jun 2010 08:52:56 -0500 Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 Please do not reply to digest emails. Subscribe normal please. Its hard to follow you if you do this. /b On Jun 29, 2010, at 11:46 AM, Ravi Kuru wrote: > Hi Steve, > > I put install the jdk 64 bit now, i don't get that error but PhoneTest.java does not working > > i compile PhoneTest.java without error and freeswitch.log i see that trying to PhoneTest but it did not work. > do you know why it is not working? ---------- Forwarded message ---------- From: David Ponzone To: freeswitch-users at lists.freeswitch.org Date: Wed, 30 Jun 2010 15:53:59 +0200 Subject: Re: [Freeswitch-users] Number of codecs offerred in SDP can you send us the output of: sofia status profile internal (of course, hide IPs and else if required). David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 30/06/2010 ? 15:46, Mark Campbell-Smith a ?crit : Nope. I have transcoding enabled. At least I have that line disabing transcoding commented out in internal.xml and I assume by default FS enables transcoding -- Sent from my Android phone with K-9 Mail. Please excuse my brevity. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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/20100630/3218473a/attachment-0001.html From sameer2k3t at gmail.com Wed Jun 30 09:02:28 2010 From: sameer2k3t at gmail.com (Sameer Khan) Date: Wed, 30 Jun 2010 21:02:28 +0500 Subject: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen In-Reply-To: References: Message-ID: Mr Giovani what was that ? Can mod_dingaling be used for multiple outgoing calls to google network like skypopen.? On Wed, Jun 30, 2010 at 7:42 PM, Milena wrote: > > If you're in a rush, you should test it yourself, if not then have some > patience and wait for someone who knows to come in and reply.... > > 2010/6/30 Sameer Khan > >> Somebody reply please >> >> >> On Tue, Jun 29, 2010 at 9:11 PM, Sameer Khan wrote: >> >>> Hi >>> Can mod_dingaling be used for multiple outgoing calls to google network >>> like skypopen.? If yes please reply >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100630/b10c5db5/attachment.html From msc at freeswitch.org Wed Jun 30 09:16:41 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Jun 2010 09:16:41 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Soon - Anthony Minessale Speaking Today Message-ID: Everyone get ready for the call: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_30 It will start in about 45 minutes from now. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/cfc98156/attachment.html From gmaruzz at celliax.org Wed Jun 30 09:19:09 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 30 Jun 2010 18:19:09 +0200 Subject: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen In-Reply-To: References: Message-ID: On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan wrote: > Mr Giovani what was that ? just jokin, nevermind -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From msc at freeswitch.org Wed Jun 30 09:45:41 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Jun 2010 09:45:41 -0700 Subject: [Freeswitch-users] Using fifo_orbit_announce For On-hook Agent In-Reply-To: References: Message-ID: On Wed, Jun 30, 2010 at 4:38 AM, afshin afzali wrote: > If your phone supports SIP update message, the actual caller-id will > be updated when the agent respond to call. > This is correct. Just two points to add: #1 - FIFO is not ACD. The call isn't transferred to the agent's phone until *after* the agent picks up, therefore there is no way to send the caller ID prior to the agent picking up #2 - Some soft phones (*cough*x-lite*cough*) do not support the display update, so be warned... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/8e9f3f95/attachment.html From msc at freeswitch.org Wed Jun 30 09:48:14 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Jun 2010 09:48:14 -0700 Subject: [Freeswitch-users] ESL: Question about BACKGROUND_JOB events In-Reply-To: <4C2B16C7.6050200@ewetel.de> References: <4C22301C.3080405@ewetel.de> <4274374276864666200@unknownmsgid> <4C29BCF6.90701@ewetel.de> <21D9FC45-0FA9-4454-97AD-010E150470FF@avgs.ca> <4C2B16C7.6050200@ewetel.de> Message-ID: 2010/6/30 Helmut Kuper > Hi, > > well, so I have to parse all BACKGROUND-JOB events in my app to find the > event for me? > > How many processes out there doing "bgapi" do you have running at once? Like bkw said, this isn't exactly a difficult task, even if you have many such processes. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/5d45491d/attachment.html From deya787 at gmail.com Wed Jun 30 09:52:53 2010 From: deya787 at gmail.com (Deya M) Date: Thu, 1 Jul 2010 02:52:53 +1000 Subject: [Freeswitch-users] Possible Mem LeakFS Message-ID: Anthony, Thanks for your reply, Rebuilding Freeswitch fixed the problem of the physical memory, Physical Memory not changing anymore, and freeswitch has fixed mem/res. D:- Jan Thanks for you Stupid Answer, but, if you cannot answer the question, would rather not contribute or reply, without getting into printf issue, which I am not sure if you understand what you are saying, or what I am saying! D:- On Thu, Jul 1, 2010 at 1:15 AM, < 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: Possible Mem LeakFS (Jan Berger) > > > ---------- Forwarded message ---------- > From: "Jan Berger" > To: > Date: Wed, 30 Jun 2010 17:14:27 +0200 > Subject: Re: [Freeswitch-users] Possible Mem LeakFS > > Deya, > > > > Stupid question ? you are not accidentally running a console app that > printf into memory ? this sounds very much like a printf issue. > > > > Jan > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Deya M > *Sent:* 30. juni 2010 16:47 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Possible Mem LeakFS > > > > Hi Anthony, > > > > I am starting with 300mb memory used, leaving the system before running fs > for 2 hours, memory is not increasing. > > As soon as I start FS, it starts going up gradually, until 2GB, and still > increasing. > > > > Is the a standard expected behaviour for allocating pools ? > > > > Bear in mind, that I am not using FS, connecting or doing any activities > with the machine. The machine is almost 100 IDLE. > > > > Thanks, > > > > D:- > > 2010/6/30 > > 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: Connect two FreeSWITCH Boxes (Milena) > 2. Re: Possible Mem LeakFS (Anthony Minessale) > 3. Re: Number of codecs offerred in SDP (Mark Campbell-Smith) > 4. Re: FreeSWITCH-users Digest, Vol 48, Issue 194 (Milena) > 5. Re: Connect two FreeSWITCH Boxes (Kane, Michael (mkane02)) > 6. Re: FreeSWITCH-users Digest, Vol 48, Issue 194 (Brian West) > 7. Re: Number of codecs offerred in SDP (David Ponzone) > > > ---------- Forwarded message ---------- > From: Milena > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 08:35:44 -0500 > Subject: Re: [Freeswitch-users] Connect two FreeSWITCH Boxes > > Hello > > > > Specifically, check the dialplan for this extension: "Calls from BoxB" or > post it so we can look at it and help you, but please don't do it here, use > pastebin!!: pastebin.freeswitch.org then send us the link. > > > > -Milena > > > > 2010/6/30 Peder > > I would guess there is an error in public on A, but you didn?t include all > of that context so I can?t say for sure. Note that it matches 10001 in > public, but when it gets to Local_Call, it just sees $1, not 10001. That s > why the match is failing. > > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 08:38:28 -0500 > Subject: Re: [Freeswitch-users] Possible Mem LeakFS > You sure about that? > > There are more bytes used in this email thread than that valgrind report > shows lost? > > We have many people using FreeSWITCH in high density scenarios and nobody > else is complaining. > > FreeSWITCH uses some memory pooling so its natural for it to retain some > memory in certain cases. > > > > On Wed, Jun 30, 2010 at 8:07 AM, Deya M wrote: > > Hi, > > > > Starting with total memory usage of 300k, I go up to 2GB after 10 hours > running FS. > > > > Just doing further tests, I got the log file, which at the end has: > > > > > > ==4573== LEAK SUMMARY: > > ==4573== definitely lost: 8,166 bytes in 26 blocks > > ==4573== indirectly lost: 284 bytes in 1 blocks > > ==4573== possibly lost: 2,794,680 bytes in 477 blocks > > ==4573== still reachable: 74,666 bytes in 2,057 blocks > > ==4573== suppressed: 0 bytes in 0 blocks > > ==4573== > > ==4573== ERROR SUMMARY: 232006 errors from 122 contexts (suppressed: 326 > from 13) > > > > Appreciate your help! > > > > > > > > http://pastebin.freeswitch.org/13314 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > > > ---------- Forwarded message ---------- > From: Mark Campbell-Smith > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 23:46:23 +1000 > Subject: Re: [Freeswitch-users] Number of codecs offerred in SDP > Nope. I have transcoding enabled. At least I have that line disabing > transcoding commented out in internal.xml and I assume by default FS enables > transcoding > > > -- > Sent from my Android phone with K-9 Mail. Please excuse my brevity. > > > > > ---------- Forwarded message ---------- > From: Milena > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 08:48:18 -0500 > Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 > > > > PASTEBIN!!! > > > > 2010/6/29 Ravi Kuru > > > > this is the freeswitch.log: > > > > ---------- Forwarded message ---------- > From: "Kane, Michael (mkane02)" > To: > Date: Wed, 30 Jun 2010 09:49:02 -0400 > Subject: Re: [Freeswitch-users] Connect two FreeSWITCH Boxes > > Sorry about that, I figured out the process of pasting after the fact. I > found the problem, as usual it was due to that thing called a human and his > fingers not coordinating with his brain. Again sorry for posting my dirty > laundry here on the list. > > > > Mike > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Milena > *Sent:* Wednesday, June 30, 2010 9:36 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Connect two FreeSWITCH Boxes > > > > Hello > > > > Specifically, check the dialplan for this extension: "Calls from BoxB" or > post it so we can look at it and help you, but please don't do it here, use > pastebin!!: pastebin.freeswitch.org then send us the link. > > > > -Milena > > > > 2010/6/30 Peder > > I would guess there is an error in public on A, but you didn?t include all > of that context so I can?t say for sure. Note that it matches 10001 in > public, but when it gets to Local_Call, it just sees $1, not 10001. That s > why the match is failing. > > > > ---------- Forwarded message ---------- > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 08:52:56 -0500 > Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 48, Issue 194 > Please do not reply to digest emails. Subscribe normal please. Its hard > to follow you if you do this. > > /b > > On Jun 29, 2010, at 11:46 AM, Ravi Kuru wrote: > > > Hi Steve, > > > > I put install the jdk 64 bit now, i don't get that error but > PhoneTest.java does not working > > > > i compile PhoneTest.java without error and freeswitch.log i see that > trying to PhoneTest but it did not work. > > do you know why it is not working? > > > > > > ---------- Forwarded message ---------- > From: David Ponzone > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 15:53:59 +0200 > Subject: Re: [Freeswitch-users] Number of codecs offerred in SDP > > can you send us the output of: > > sofia status profile internal > > (of course, hide IPs and else if required). > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > *www.ipeva.fr* - *www.ipeva-studio.com* > > > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message > s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de > ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.* > > * * > > > > > > > > Le 30/06/2010 ? 15:46, Mark Campbell-Smith a ?crit : > > > > Nope. I have transcoding enabled. At least I have that line disabing > transcoding commented out in internal.xml and I assume by default FS enables > transcoding > > > -- > Sent from my Android phone with K-9 Mail. Please excuse my brevity. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/c98e58b6/attachment-0001.html From msc at freeswitch.org Wed Jun 30 10:45:31 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Jun 2010 10:45:31 -0700 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: Can you supply a console log of these calls? -MC On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. wrote: > Dear Anthony, > I've tried using the group_confirm_cancel_timeout as per the discussion > we had in IRC. You wanted to used it as part of dial string and not as a > channel variable. > But, It doesn't work for me. > > Here is how I've given the commands and the script I've executed. Even when > I give group_confirm_cancel_timeout, the callee's leg is getting > disconnected after legtimeout. > > > connect > > sendmsg > call-command: execute > execute-app-name:answer > > sendmsg > call-command: execute > execute-app-name: set > execute-app-arg: group_confirm_key=exec > > sendmsg > call-command: execute > execute-app-name: set > execute-app-arg: group_confirm_file=perl /root/bridge.pl > > sendmsg > call-command: execute > execute-app-name: bridge > execute-app-arg: {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 > > > > bridge.pl: > #!/usr/bin/perl > use freeswitch; > > our $session; > freeswitch::consoleLog("info","Goint to get the digits"); > # To simulate the scenario I used sleep here. > sleep(30); > 1; > > Kindly tell me whats wrong in the above. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/750949fc/attachment.html From msc at freeswitch.org Wed Jun 30 11:27:42 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Jun 2010 11:27:42 -0700 Subject: [Freeswitch-users] ClueCon News Update - Featured Speakers, More Giveaways Message-ID: As ClueCon MMX gets closer we have more good news to share with everyone. First of all, we are pleased to announce that Philip Zimmermann, inventor of PGP, will be returning to ClueCon this year! We look forward to his talk: "Secure VoIP Explained - Why You Need Secure VoIP and the Best Way to Get It." Our other featured speakers include: * Anthony Minessale, author of the FreeSWITCH project: Keynote Address * Daniel-Constantin Mierla, co-founder and core developer of Kamailio: Blending FreeSWITCH and Kamailio to Build Large Unified Communication Platforms * Bogdan-Andrei Iancu, founder of OpenSIPS: OpenSIPS Load-Balancing For A FreeSWITCH Server Cluster * Matt Florell, author of the ViciDial Call Center Suite: Building a Scalable Hosted Call Center Platform * Moises Silva, software engineer, Sangoma Technologies: The FreeTDM API * Darren Schreiber, founder of the 2600Hz project and former lead developer of FreePBX V3: Distributed Cloud Telephony With 2600Hz We certainly look forward to these presentations and more. The complete ClueCon schedule is available here. ClueCon has a new sponsor: Meraki. Meraki will be supplying WiFi access points and the necessary resources to deploy and maintain them throughout the conference. Meraki will be making sure that all of our guests have a smooth and pleasant experience with their network connectivity. In other news, Sangoma Technologies, a diamond sponsor of this year's event, will be giving away several TDM cards: * A200 4-port analog card with 2 FXO and 2 FXS ports * A101DE single port T1/E1 Both of these cards are PCI Express and have built in echo cancellation. Sangoma will be giving away four of each - that's eight TDM cards total! This is in addition to the two iPads and the Macbook Pro. The engraving of these items will be done soon - watch for news and pictures. Of course, we probably have other goodies to give away as well... We also have a hotel alert: The week of August 2, 2010 is going to be a busy one for Chicago hotels. A large musical event is being held on Friday August 6 and local hotels are starting to fill up. Be sure to book your room at the Trump by Tuesday July 13 so that you can secure the rate of $225 per night. Come visit our Web site or call 877.742.CLUE (2583) for more information. See you in August! The ClueCon Team http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/96a7b6be/attachment.html From infos at madovsky.org Wed Jun 30 11:57:50 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 30 Jun 2010 14:57:50 -0400 Subject: [Freeswitch-users] ClueCon News Update - Featured Speakers, More Giveaways References: Message-ID: No problem for the Hotel, I will sleep in my car ;) We also have a hotel alert: The week of August 2, 2010 is going to be a busy one for Chicago hotels. A large musical event is being held on Friday August 6 and local hotels are starting to fill up. Be sure to book your room at the Trump by Tuesday July 13 so that you can secure the rate of $225 per night. Come visit our Web site or call 877.742.CLUE (2583) for more information. See you in August! The ClueCon Team http://www.cluecon.com 877.742.CLUE ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/5e0ca46d/attachment.html From javieraristizabal at gmail.com Wed Jun 30 13:19:13 2010 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Wed, 30 Jun 2010 15:19:13 -0500 Subject: [Freeswitch-users] ClueCon News Update - Featured Speakers, More Giveaways In-Reply-To: References: Message-ID: That's the spirit Franck!! On Wed, Jun 30, 2010 at 1:57 PM, Madovsky wrote: > No problem for the Hotel, I will sleep in my car ;) > > > We also have a hotel alert: The week of August 2, 2010 is going to be a > busy one for Chicago hotels. A large musical event is being held on Friday > August 6 and local hotels are starting to fill up. Be sure to book your room > at the Trump by Tuesday July 13 so that you can secure the rate of $225 per > night. Come visit our Web site or call 877.742.CLUE (2583) for more > information. > > See you in August! > > The ClueCon Team > http://www.cluecon.com > 877.742.CLUE > > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > -- Javier Aristiz?bal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/d4a8ec2d/attachment.html From sameer2k3t at gmail.com Wed Jun 30 13:39:35 2010 From: sameer2k3t at gmail.com (Sameer Khan) Date: Thu, 1 Jul 2010 01:39:35 +0500 Subject: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen In-Reply-To: References: Message-ID: no problem. but did't expect from the creator of mod_skypopen, the wonder i would say On Wed, Jun 30, 2010 at 9:19 PM, Giovanni Maruzzelli wrote: > On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan wrote: > > Mr Giovani what was that ? > > just jokin, nevermind > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/0ea44901/attachment.html From infos at madovsky.org Wed Jun 30 13:39:59 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 30 Jun 2010 16:39:59 -0400 Subject: [Freeswitch-users] ClueCon News Update - Featured Speakers, More Giveaways References: Message-ID: <35ED1D17769E40868C9B561444E3197A@MOBILEE1705> God, I even had no car !! :D ----- Original Message ----- From: Javier Aristiz?bal To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, June 30, 2010 4:19 PM Subject: Re: [Freeswitch-users] ClueCon News Update - Featured Speakers,More Giveaways That's the spirit Franck!! On Wed, Jun 30, 2010 at 1:57 PM, Madovsky wrote: No problem for the Hotel, I will sleep in my car ;) We also have a hotel alert: The week of August 2, 2010 is going to be a busy one for Chicago hotels. A large musical event is being held on Friday August 6 and local hotels are starting to fill up. Be sure to book your room at the Trump by Tuesday July 13 so that you can secure the rate of $225 per night. Come visit our Web site or call 877.742.CLUE (2583) for more information. See you in August! The ClueCon Team http://www.cluecon.com 877.742.CLUE -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.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 -- Javier Aristiz?bal ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100630/c309f05a/attachment-0001.html From sos at sokhapkin.dyndns.org Wed Jun 30 13:50:21 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 30 Jun 2010 16:50:21 -0400 Subject: [Freeswitch-users] =?iso-8859-15?q?ClueCon_News_Update_-_Featured?= =?iso-8859-15?q?_Speakers=2C_More_=09Giveaways?= In-Reply-To: <35ED1D17769E40868C9B561444E3197A@MOBILEE1705> References: <35ED1D17769E40868C9B561444E3197A@MOBILEE1705> Message-ID: <201006301650.21940.sos@sokhapkin.dyndns.org> Hmm, how do you feed BP then??? BP offers a nice promotion here in US - fill up for free with whatever you can skim in the Gulf :-\ On Wednesday 30 June 2010, Madovsky wrote: > God, I even had no car !! :D > ----- Original Message ----- > From: Javier Aristiz?bal > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, June 30, 2010 4:19 PM > Subject: Re: [Freeswitch-users] ClueCon News Update - Featured > Speakers,More Giveaways > > > That's the spirit Franck!! > > > On Wed, Jun 30, 2010 at 1:57 PM, Madovsky wrote: > > No problem for the Hotel, I will sleep in my car ;) > > We also have a hotel alert: The week of August 2, 2010 is going to be > a busy one for Chicago hotels. A large musical event is being held on > Friday August 6 and local hotels are starting to fill up. Be sure to book > your room at the Trump by Tuesday July 13 so that you can secure the rate > of $225 per night. Come visit our Web site or call 877.742.CLUE (2583) for > more information. > > See you in August! > > The ClueCon Team > http://www.cluecon.com > 877.742.CLUE > > > > > > -------------------------------------------------------------------------- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > > > > > > > > > --------------------------------------------------------------------------- > --- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 infos at madovsky.org Wed Jun 30 13:57:27 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 30 Jun 2010 16:57:27 -0400 Subject: [Freeswitch-users] ClueCon News Update - Featured Speakers, More Giveaways References: <35ED1D17769E40868C9B561444E3197A@MOBILEE1705> <201006301650.21940.sos@sokhapkin.dyndns.org> Message-ID: <7E8604B1BAA5419F96BF114E2F18E696@MOBILEE1705> maybe they will need FreeSWIITCH to prevent another further catastroph with a better communication ! :) ----- Original Message ----- From: "Sergey Okhapkin" To: Sent: Wednesday, June 30, 2010 4:50 PM Subject: Re: [Freeswitch-users] ClueCon News Update - Featured Speakers, More Giveaways Hmm, how do you feed BP then??? BP offers a nice promotion here in US - fill up for free with whatever you can skim in the Gulf :-\ On Wednesday 30 June 2010, Madovsky wrote: > God, I even had no car !! :D > ----- Original Message ----- > From: Javier Aristiz?bal > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, June 30, 2010 4:19 PM > Subject: Re: [Freeswitch-users] ClueCon News Update - Featured > Speakers,More Giveaways > > > That's the spirit Franck!! > > > On Wed, Jun 30, 2010 at 1:57 PM, Madovsky wrote: > > No problem for the Hotel, I will sleep in my car ;) > > We also have a hotel alert: The week of August 2, 2010 is going to > be > a busy one for Chicago hotels. A large musical event is being held on > Friday August 6 and local hotels are starting to fill up. Be sure to book > your room at the Trump by Tuesday July 13 so that you can secure the rate > of $225 per night. Come visit our Web site or call 877.742.CLUE (2583) > for > more information. > > See you in August! > > The ClueCon Team > http://www.cluecon.com > 877.742.CLUE > > > > > > -------------------------------------------------------------------------- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > > > > > > > > > --------------------------------------------------------------------------- > --- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 From mcampbellsmith at gmail.com Wed Jun 30 15:15:55 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 1 Jul 2010 08:15:55 +1000 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> <7D2C6429-EB5F-4720-90EE-6B0C0410A34A@gmail.com> <14709fd5-c2ce-4627-abd9-e33bcc8b0d03@email.android.com> Message-ID: Sure David. thanks for helping me with this. I have not included the registrations, as I didn't think that was necessary .... freeswitch at internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.1.120 Ext-RTP-IP 124.xx.xx.xx SIP-IP 192.168.1.120 Ext-SIP-IP 124.xx.xx.xx URL sip:mod_sofia at 192.168.1.120:5060 BIND-URL sip:mod_sofia at 192.168.1.120:5060 TLS-URL sip:mod_sofia at 192.168.1.120:442 TLS-BIND-URL sips:mod_sofia at 192.168.1.120:442;transport=tls HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G729,PCMU,GSM CODECS OUT G729,PCMU,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 40 FAILED-CALLS-IN 8 CALLS-OUT 18 FAILED-CALLS-OUT 11 Registrations: 2010/6/30 David Ponzone : > can you send us the output of: > sofia status profile internal > (of course, hide IPs and else if required). > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 30/06/2010 ? 15:46, Mark Campbell-Smith a ?crit : > > Nope. I have transcoding enabled. At least I have that line disabing > transcoding commented out in internal.xml and I assume by default FS enables > transcoding > > > -- > Sent from my Android phone with K-9 Mail. Please excuse my brevity. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > From anthony.minessale at gmail.com Wed Jun 30 15:39:08 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Jun 2010 17:39:08 -0500 Subject: [Freeswitch-users] Number of codecs offerred in SDP In-Reply-To: References: <699019ED-B294-4546-AE7D-26E2A59C84FA@gmail.com> <7D2C6429-EB5F-4720-90EE-6B0C0410A34A@gmail.com> <14709fd5-c2ce-4627-abd9-e33bcc8b0d03@email.android.com> Message-ID: in your dialplan add {absolute_codec_string=GSM} to your url On Wed, Jun 30, 2010 at 5:15 PM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Sure David. thanks for helping me with this. I have not included the > registrations, as I didn't think that was necessary .... > > freeswitch at internal> sofia status profile internal > > ================================================================================================= > Name internal > Domain Name N/A > Auto-NAT true > DBName sofia_reg_internal > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_from > RTP-IP 192.168.1.120 > Ext-RTP-IP 124.xx.xx.xx > SIP-IP 192.168.1.120 > Ext-SIP-IP 124.xx.xx.xx > URL sip:mod_sofia at 192.168.1.120:5060 > BIND-URL sip:mod_sofia at 192.168.1.120:5060 > TLS-URL sip:mod_sofia at 192.168.1.120:442 > TLS-BIND-URL sips:mod_sofia at 192.168.1.120:442;transport=tls > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS IN G729,PCMU,GSM > CODECS OUT G729,PCMU,GSM > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT true > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 40 > FAILED-CALLS-IN 8 > CALLS-OUT 18 > FAILED-CALLS-OUT 11 > > Registrations: > > 2010/6/30 David Ponzone : > > can you send us the output of: > > sofia status profile internal > > (of course, hide IPs and else if required). > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > > l'intention exclusive de ses destinataires. Toute utilisation ou > diffusion > > non autoris?e est interdite. Tout message ?lectronique est susceptible > > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message > s'il > > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > > > > Le 30/06/2010 ? 15:46, Mark Campbell-Smith a ?crit : > > > > Nope. I have transcoding enabled. At least I have that line disabing > > transcoding commented out in internal.xml and I assume by default FS > enables > > transcoding > > > > > > -- > > Sent from my Android phone with K-9 Mail. Please excuse my brevity. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100630/42fb6b8e/attachment-0001.html From anthony.minessale at gmail.com Wed Jun 30 15:40:27 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Jun 2010 17:40:27 -0500 Subject: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen In-Reply-To: References: Message-ID: If he didn't like to joke, he'd never survive in telephony. On Wed, Jun 30, 2010 at 3:39 PM, Sameer Khan wrote: > no problem. but did't expect from the creator of mod_skypopen, the wonder i > would say > > > On Wed, Jun 30, 2010 at 9:19 PM, Giovanni Maruzzelli wrote: > >> On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan >> wrote: >> > Mr Giovani what was that ? >> >> just jokin, nevermind >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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 > > -- 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/20100630/b3bca842/attachment.html From xengelpublicx at gmail.com Wed Jun 30 16:29:09 2010 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Thu, 1 Jul 2010 03:29:09 +0400 Subject: [Freeswitch-users] continue_on_fail and hangup_after_bridge with transfer Message-ID: I'm trying to make a dialplan: The logic of his work: if unavailable gateway to the next. if not available all the gateway to go to the extension transfer. A problem in If we get a code busy here, it is not satisfied hangup_after_bridge. The call goes to the transfer. Why is this so? trace: http://pastebin.freeswitch.org/13244 -- Best regards, Vladimir Elizarov From fs-list at communicatefreely.net Wed Jun 30 19:45:19 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Wed, 30 Jun 2010 22:45:19 -0400 Subject: [Freeswitch-users] SIP header on only one fork of a bridge In-Reply-To: References: <4C28EDB8.6080609@communicatefreely.net> Message-ID: <4C2C013F.6070505@communicatefreely.net> Thanks all, I read that bit, but didn't fully understand it's implications. I'll give that a go. -Tim Steven Ayre wrote: > data="{this_is_global=true}[this_is_gw1_only=true]sofia/gateway/gw1/$1|[this_is_gw2_only=true]sofia/gateway/gw2/$1"/> > > > > On 28 June 2010 19:45, Tim St. Pierre > wrote: > > Hello list, > > I would like to bridge a call to multiple SIP endpoints, but add > different headers to each. > > I'm not entirely sure how to do this. I have no problem exporting a > SIP header that does what I > want for one destination, but I'm not sure how to set it for two. > > My application is that I want two IP phones to ring - one with the > internal ring-ring splash, the > others with a group-answer (single ring, then lamp flash only), for > administrative assistants, etc. > > How do I export different variables to each branch? > > Thanks! > > -Tim > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 From deya787 at gmail.com Wed Jun 30 20:19:51 2010 From: deya787 at gmail.com (Deya M) Date: Thu, 1 Jul 2010 06:19:51 +0300 Subject: [Freeswitch-users] Default Dial Plan: action application bridge Message-ID: Hi, In the default dial plan, with two extensions defined, 1000 and 1001, when I call from 1000 to 1001, I always get Voicemail, using the default config files: conf/dialplan/default.xml I changed the following from : * * TO ** and it did work! Not sure if the first / default one, * *should work, but something is missing ? New: Old: :-D -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/266c5ea5/attachment.html From anthony.minessale at gmail.com Wed Jun 30 20:37:15 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Jun 2010 22:37:15 -0500 Subject: [Freeswitch-users] Default Dial Plan: action application bridge In-Reply-To: References: Message-ID: you must have changed more than you think. you might want to diff you configs against the in-tree ones. On Wed, Jun 30, 2010 at 10:19 PM, Deya M wrote: > Hi, > > In the default dial plan, with two extensions defined, 1000 and 1001, when > I call from 1000 to 1001, I always get Voicemail, using the default config > files: conf/dialplan/default.xml > > I changed the following from : > > * * > TO > * data="sofia/internal/${dialed_extension}%${domain_name}"/>* > > and it did work! Not sure if the first / default one, * application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should > work, but something is missing ? > > > New: > > > > > > > > > > > > > > > > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > data="sofia/internal/${dialed_extension}%${domain_name}"/> > > > > > > > > Old: > > > > > > > > > > > > > > > > > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > > > > > > > > :-D > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100630/9b862b52/attachment-0001.html From anthony.minessale at gmail.com Wed Jun 30 20:41:19 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Jun 2010 22:41:19 -0500 Subject: [Freeswitch-users] calls ending with MEDIA_TIMEOUT In-Reply-To: <23029141.3084.1277908748262.JavaMail.daniel@radio> References: <23029141.3084.1277908748262.JavaMail.daniel@radio> Message-ID: you neet to get pcaps of the calls and look at the rtp and sip going on. On Wed, Jun 30, 2010 at 9:39 AM, Dan wrote: > Thanks for your response, I put everything up on pastebin > http://pastebin.freeswitch.org/13322 . The application in question is > actually javascript, I'm using lua in production but was switching to the > posted js version with the upgrade. > > Now that I posted it i realized I have > > > > in my dial plan, I believed I used it in the older version to get around > some dtmf issues or choppy playback (can't remember), not sure if this could > be part of the issue (although it works fine in the production version I'm > running) > > > So I pulled one of the recordings that hung up after 4 minutes, but was > only 24 seconds long, it sounded fine (but obviously too short). But > another one that dropped after 5 minutes and only 19 seconds in length was > very choppy and included short spurts of audio from parts of the call that > were much longer then 19 seconds. > ------------------------------ > *From: *"Anthony Minessale" > *To: *freeswitch-users at lists.freeswitch.org > *Sent: *Tuesday, June 29, 2010 12:54:21 PM > *Subject: *Re: [Freeswitch-users] calls ending with MEDIA_TIMEOUT > > > it's not 100% accurate in the media timeout. > It would be too expensive to use actual timers, it uses the number of > samples you should be getting from rtp > and a number of loops where no media was received. > > Migrating from svn 13000 range to GIT is a big step and you may have to > adjust to some new behaviors. > media_timeout may not even have existed that long ago I don't recall. > > If you don't need media timeouts turn off the param or turn it up to > longer. > > > On Tue, Jun 29, 2010 at 1:09 PM, Michael Collins wrote: > >> Pastebin your dialplan and the lua script for starters. Also, is it the >> 5300 that is responding with the media timeout? >> -MC >> >> On Tue, Jun 29, 2010 at 10:15 AM, Dan wrote: >> >>> Hi guys, I have been running two freeswitch boxes (13754M) that answer >>> calls from a cisco 5300 (both on the same network) and records them to disk >>> with a small lua application. This has been working well for the past few >>> months. I decided to upgrade one of them to trunk ( git-3fbd9e2 2010-06-11 >>> 11-08-51 -0500 ) and have run into a problem. Some calls will fail with a >>> MEDIA_TIMEOUT after a few minutes, the time it takes to fail ranges from 4 >>> minutes to 10 minutes, I don't have a full sip trace or pcap dump yet, I >>> reverted back to the old freeswitch version (on the same hardware) and have >>> not been able to reproduce it in a test environment yet ( I continue to >>> try). Below are the relevant lines from the log files for one of the >>> calls: >>> >>> 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2257 (sofia/external/ >>> nobody at 192.168.21.4) Callstate Change ACTIVE -> HANGUP >>> 2010-06-23 07:42:19.033466 [NOTICE] mod_sofia.c:884 Hangup >>> sofia/external/nobody at 192.168.21.4 [CS_EXECUTE] [MEDIA_TIMEOUT] >>> 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2273 Send signal >>> sofia/external/nobody at 192.168.21.4 [KILL] >>> 2010-06-23 07:42:19.033466 [DEBUG] switch_core_session.c:1023 Send signal >>> sofia/external/nobody at 192.168.21.4 [BREAK] >>> 2010-06-23 07:42:19.033466 [DEBUG] switch_core_codec.c:146 >>> sofia/external/nobody at 192.168.21.4 Restore previous codec PCMU:0. >>> >>> My configuration is bone stock, so the rtp timeout value is at 300, but >>> I have some calls that have lasted only 4 minutes. One other piece of >>> information is that on one of the recordings that was hung up after 4 >>> minutes and 17 seconds the recorded file was only 24 seconds long (it >>> stopped recording after the first 24 seconds) , so I'm assuming freeswitch >>> did not think there were any rtp packets to record. >>> >>> Any ideas on where to start debugging this? I have setup a new >>> freeswitch box connected to the same 5300 to reproduce, but have not been >>> able to generate the call volume ( there where around 30 calls being >>> recorded) yet, but I'm working on it. >>> >>> Thanks! >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > > -- > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.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 > > -- 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/20100630/ab278d40/attachment.html From anthony.minessale at gmail.com Wed Jun 30 20:52:33 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Jun 2010 22:52:33 -0500 Subject: [Freeswitch-users] Using fifo_orbit_announce For On-hook Agent In-Reply-To: References: Message-ID: try latest GIT, there is new outbound_strategy param which defaults to "ringall" which will be what you want. The bad part is now you have to wait for each call to be answered before you can make another call just so you can send the caller id in the initial call. *shrug* that's how everybody wants it, then go for it its in tree.... On Wed, Jun 30, 2010 at 11:45 AM, Michael Collins wrote: > > > On Wed, Jun 30, 2010 at 4:38 AM, afshin afzali wrote: > >> If your phone supports SIP update message, the actual caller-id will >> be updated when the agent respond to call. >> > > This is correct. Just two points to add: > > #1 - FIFO is not ACD. The call isn't transferred to the agent's phone until > *after* the agent picks up, therefore there is no way to send the caller ID > prior to the agent picking up > > #2 - Some soft phones (*cough*x-lite*cough*) do not support the display > update, so be warned... > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20100630/2c518bc4/attachment.html From deya787 at gmail.com Wed Jun 30 21:47:44 2010 From: deya787 at gmail.com (Deya M) Date: Thu, 1 Jul 2010 14:47:44 +1000 Subject: [Freeswitch-users] Create a external FXO Trunk - A200 Message-ID: Hi, San A200 FXO is woking with wanpipe, and freeswitch detects and init : 2010-07-01 09:07:15.427539 [NOTICE] switch_loadable_module.c:145 Adding Endpoint 'loopback' 2010-07-01 09:07:15.428061 [DEBUG] zap_config.c:56 Configuration file is /usr/local/freeswitch/conf/modules.conf. 2010-07-01 09:07:15.428125 [NOTICE] zap_io.c:3058 Modules configured: 1 2010-07-01 09:07:15.428137 [DEBUG] zap_config.c:56 Configuration file is /usr/local/freeswitch/conf/openzap.conf. 2010-07-01 09:07:15.428161 [DEBUG] zap_io.c:2588 found config for span 2010-07-01 09:07:15.428404 [INFO] zap_io.c:2859 Loading IO from /usr/local/freeswitch/mod/ozmod_wanpipe.so [wanpipe] 2010-07-01 09:07:15.428418 [DEBUG] zap_config.c:56 Configuration file is /usr/local/freeswitch/conf/wanpipe.conf. 2010-07-01 09:07:15.428467 [INFO] zap_io.c:2605 auto-loaded 'wanpipe' 2010-07-01 09:07:15.428494 [DEBUG] zap_io.c:2626 created span 1 (FXO) of type wanpipe 2010-07-01 09:07:15.428503 [DEBUG] zap_io.c:2639 span 1 [name]=[OpenZAP] 2010-07-01 09:07:15.428510 [DEBUG] zap_io.c:2639 span 1 [fxo-channel]=[1:1] 2010-07-01 09:07:15.428518 [DEBUG] zap_io.c:2668 setting trunk type to 'FXO' start(KEWL) 2010-07-01 09:07:15.428570 [INFO] ozmod_wanpipe.c:330 configuring device s1c1 as OpenZAP device 1:1 fd:41 DTMF: hardware 2010-07-01 09:07:15.428587 [DEBUG] zap_io.c:2639 span 1 [fxo-channel]=[1:2] 2010-07-01 09:07:15.428632 [INFO] ozmod_wanpipe.c:330 configuring device s1c2 as OpenZAP device 1:2 fd:42 DTMF: hardware 2010-07-01 09:07:15.428643 [DEBUG] zap_io.c:2639 span 1 [fxo-channel]=[1:3] 2010-07-01 09:07:15.428695 [INFO] ozmod_wanpipe.c:330 configuring device s1c3 as OpenZAP device 1:3 fd:43 DTMF: hardware 2010-07-01 09:07:15.428707 [DEBUG] zap_io.c:2639 span 1 [fxo-channel]=[1:4] 2010-07-01 09:07:15.428748 [INFO] ozmod_wanpipe.c:330 configuring device s1c4 as OpenZAP device 1:4 fd:44 DTMF: hardware 2010-07-01 09:07:15.428774 [INFO] zap_io.c:2783 Configured 4 channel(s) 2010-07-01 09:07:15.428920 [INFO] zap_io.c:2876 Loading SIG from /usr/local/freeswitch/mod/ozmod_analog.so 2010-07-01 09:07:15.428932 [INFO] zap_io.c:2992 auto-loaded 'analog' 2010-07-01 09:07:15.428943 [DEBUG] ozmod_analog.c:184 Enabling call waiting for channel 1:1 2010-07-01 09:07:15.428957 [DEBUG] ozmod_analog.c:184 Enabling call waiting for channel 1:2 2010-07-01 09:07:15.428963 [DEBUG] ozmod_analog.c:184 Enabling call waiting for channel 1:3 2010-07-01 09:07:15.428970 [DEBUG] ozmod_analog.c:184 Enabling call waiting for channel 1:4 2010-07-01 09:07:15.428977 [DEBUG] zap_config.c:56 Configuration file is /usr/local/freeswitch/conf/tones.conf. 2010-07-01 09:07:15.429003 [DEBUG] zap_io.c:551 added tone generation [dial] = [v=-7;%(1000,0,350,440)] 2010-07-01 09:07:15.429028 [DEBUG] zap_io.c:549 added tone detect [dial] = [350,440] 2010-07-01 09:07:15.429037 [DEBUG] zap_io.c:551 added tone generation [ring] = [v=-7;%(2000,4000,440,480)] 2010-07-01 09:07:15.429045 [DEBUG] zap_io.c:549 added tone detect [ring] = [440,480] 2010-07-01 09:07:15.429053 [DEBUG] zap_io.c:551 added tone generation [busy] = [v=-7;%(500,500,480,620)] 2010-07-01 09:07:15.429062 [DEBUG] zap_io.c:549 added tone detect [busy] = [480,620] 2010-07-01 09:07:15.429070 [DEBUG] zap_io.c:551 added tone generation [attn] = [v=0;%(100,100,1400,2060,2450,2600)] 2010-07-01 09:07:15.429078 [DEBUG] zap_io.c:549 added tone detect [attn] = [1400,2060,2450,2600] 2010-07-01 09:07:15.429086 [DEBUG] zap_io.c:551 added tone generation [callwaiting-sas] = [v=0;%(300,0,440)] 2010-07-01 09:07:15.429095 [DEBUG] zap_io.c:549 added tone detect [callwaiting-sas] = [440] 2010-07-01 09:07:15.429104 [DEBUG] zap_io.c:551 added tone generation [callwaiting-cas] = [v=0;%(80,0,2750,2130)] 2010-07-01 09:07:15.429113 [DEBUG] zap_io.c:549 added tone detect [callwaiting-cas] = [2750,2130] 2010-07-01 09:07:15.429125 [DEBUG] zap_io.c:549 added tone detect [fail1] = [913.8] 2010-07-01 09:07:15.429134 [DEBUG] zap_io.c:549 added tone detect [fail2] = [1370.6] 2010-07-01 09:07:15.429141 [DEBUG] zap_io.c:549 added tone detect [fail3] = [1776.7] 2010-07-01 09:07:15.429219 [NOTICE] switch_loadable_module.c:145 Adding Endpoint 'openzap' 2010-07-01 09:07:15.429254 [NOTICE] switch_loadable_module.c:251 Adding Application 'disable_ec' 2010-07-01 09:07:15.429290 [NOTICE] switch_loadable_module.c:251 Adding Application 'disable_dtmf' 2010-07-01 09:07:15.429332 [NOTICE] switch_loadable_module.c:251 Adding Application 'enable_dtmf' 2010-07-01 09:07:15.429381 [NOTICE] switch_loadable_module.c:273 Adding API Function 'oz' openzap.conf: [span wanpipe FXO] name => OpenZAP fxo-channel => 1:1 fxo-channel => 1:2 fxo-channel => 1:3 fxo-channel => 1:4 openzap.conf.xml: configuration name="openzap.conf" description="OpenZAP Configuration"> wanrouter status Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1 | N/A | A200/A400/B600/B700| 177 | 4 | 1 | N/A | 0 | Wanrouter Status: Device name | Protocol | Station | Status | wanpipe1 | A-ANALOG | N/A | Connected | My questions, is how can I use this trunk, line 1-4, to dial ? I couldn't find a similar example using A200 Analog, could you please send me an example? Thanks, -:D -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/be384069/attachment-0001.html From deya787 at gmail.com Wed Jun 30 21:49:26 2010 From: deya787 at gmail.com (Deya M) Date: Thu, 1 Jul 2010 14:49:26 +1000 Subject: [Freeswitch-users] Default Dial Plan: action application bridge Message-ID: I generated samples, and started from scratch. I also made a copy of the conf after installation, rm it, and copied the original conf before testing! -:D On Thu, Jul 1, 2010 at 1:37 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: Query !! freeswitch, dingaling, jingle, skypopen > (Anthony Minessale) > 2. continue_on_fail and hangup_after_bridge with transfer > (Vladimir Elizarov) > 3. Re: SIP header on only one fork of a bridge (Tim St. Pierre) > 4. Default Dial Plan: action application bridge (Deya M) > 5. Re: Default Dial Plan: action application bridge > (Anthony Minessale) > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 17:40:27 -0500 > Subject: Re: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, > skypopen > If he didn't like to joke, he'd never survive in telephony. > > > On Wed, Jun 30, 2010 at 3:39 PM, Sameer Khan wrote: > >> no problem. but did't expect from the creator of mod_skypopen, the wonder >> i would say >> >> >> On Wed, Jun 30, 2010 at 9:19 PM, Giovanni Maruzzelli > > wrote: >> >>> On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan >>> wrote: >>> > Mr Giovani what was that ? >>> >>> just jokin, nevermind >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.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 >> >> > > > -- > 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 > > > ---------- Forwarded message ---------- > From: Vladimir Elizarov > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 1 Jul 2010 03:29:09 +0400 > Subject: [Freeswitch-users] continue_on_fail and hangup_after_bridge with > transfer > I'm trying to make a dialplan: > > > expression="^(^([0-9]{10})$|^([0-9]{11})$|^([0-9]{12})$|^\+([0-9]{11})$)$"> > data="hangup_after_bridge=true"/> > application="set"data="continue_on_fail=NO_ROUTE_DESTINATION"/> > > > > break="on-true"> > data="dtmf=WWWWWWWWWWWWWWWWWWWWW111222#WWWWWW89$2#@100"/> > > data="sofia/gateway/gw1/89$2|sofia/gateway/gw2/89$2|sofia/gateway/gw3/89$2"/> > > > > > > > data="{execute_on_answer=send_dtmf\s${dtmf}}sofia/internal/${ > real_dialed_number}@192.168.50.53:5061"/> > > > > The logic of his work: if unavailable gateway to the next. if not > available all the gateway to go to the extension transfer. > A problem in If we get a code busy here, it is not satisfied > hangup_after_bridge. The call goes to the transfer. Why is this so? > > trace: > http://pastebin.freeswitch.org/13244 > > -- > Best regards, Vladimir Elizarov > > > > > ---------- Forwarded message ---------- > From: "Tim St. Pierre" > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 22:45:19 -0400 > Subject: Re: [Freeswitch-users] SIP header on only one fork of a bridge > Thanks all, > > I read that bit, but didn't fully understand it's implications. > > I'll give that a go. > > -Tim > > Steven Ayre wrote: > > > > data="{this_is_global=true}[this_is_gw1_only=true]sofia/gateway/gw1/$1|[this_is_gw2_only=true]sofia/gateway/gw2/$1"/> > > > > > > > > On 28 June 2010 19:45, Tim St. Pierre > > wrote: > > > > Hello list, > > > > I would like to bridge a call to multiple SIP endpoints, but add > > different headers to each. > > > > I'm not entirely sure how to do this. I have no problem exporting a > > SIP header that does what I > > want for one destination, but I'm not sure how to set it for two. > > > > My application is that I want two IP phones to ring - one with the > > internal ring-ring splash, the > > others with a group-answer (single ring, then lamp flash only), for > > administrative assistants, etc. > > > > How do I export different variables to each branch? > > > > Thanks! > > > > -Tim > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.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 > > > > > > ---------- Forwarded message ---------- > From: Deya M > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 1 Jul 2010 06:19:51 +0300 > Subject: [Freeswitch-users] Default Dial Plan: action application bridge > Hi, > > In the default dial plan, with two extensions defined, 1000 and 1001, when > I call from 1000 to 1001, I always get Voicemail, using the default config > files: conf/dialplan/default.xml > > I changed the following from : > > * * > TO > * data="sofia/internal/${dialed_extension}%${domain_name}"/>* > > and it did work! Not sure if the first / default one, * application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should > work, but something is missing ? > > > New: > > > > > > > > > > > > > > > > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > data="sofia/internal/${dialed_extension}%${domain_name}"/> > > > > > > > > Old: > > > > > > > > > > > > > > > > > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > > > > > > > > :-D > > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Jun 2010 22:37:15 -0500 > Subject: Re: [Freeswitch-users] Default Dial Plan: action application > bridge > you must have changed more than you think. > > you might want to diff you configs against the in-tree ones. > > > On Wed, Jun 30, 2010 at 10:19 PM, Deya M wrote: > >> Hi, >> >> In the default dial plan, with two extensions defined, 1000 and 1001, when >> I call from 1000 to 1001, I always get Voicemail, using the default config >> files: conf/dialplan/default.xml >> >> I changed the following from : >> >> * * >> TO >> *> data="sofia/internal/${dialed_extension}%${domain_name}"/>* >> >> and it did work! Not sure if the first / default one, * > application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should >> work, but something is missing ? >> >> >> New: >> >> >> >> >> >> >> >> >> >> > data="transfer_ringback=$${hold_music}"/> >> >> >> >> >> >> > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >> var callgroup)}"/> >> >> > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> > data="sofia/internal/${dialed_extension}%${domain_name}"/> >> >> >> >> >> >> >> >> Old: >> >> >> >> >> >> >> >> >> >> >> > data="transfer_ringback=$${hold_music}"/> >> >> >> >> >> >> > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >> var callgroup)}"/> >> >> > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> >> >> >> >> >> >> >> :-D >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/b09db79e/attachment-0001.html From deya787 at gmail.com Wed Jun 30 23:16:36 2010 From: deya787 at gmail.com (Deya M) Date: Thu, 1 Jul 2010 16:16:36 +1000 Subject: [Freeswitch-users] create a external FXO Trunk - A200 Message-ID: Hi, Managed to ge I added the following to the dialplan: No ring. I get the following: 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:403 Set codec PCMA 20ms 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1317 Connect outbound channel OpenZAP/1:4/ 2010-07-01 11:07:04.627394 [NOTICE] switch_channel.c:776 New Channel OpenZAP/1:4/ [3ca74e5b-7e26-4a0a-b5f7-72d7a2f33fbf] 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1331 (OpenZAP/1:4/) State Change CS_NEW -> CS_INIT 2010-07-01 11:07:04.627394 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:59 Changing state on 1:4 from DOWN to DIALING 2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:293 ANALOG CHANNEL thread starting. 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_INIT 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT 2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:431 (OpenZAP/1:4/) State Change CS_INIT -> CS_ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT going to sleep 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_channel.c:1471 (OpenZAP/1:4/) Callstate Change DOWN -> RINGING 2010-07-01 11:07:04.627394 [DEBUG] ozmod_wanpipe.c:608 Enabled DTMF events on chan 1:4 2010-07-01 11:07:04.628392 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DIALING 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING 2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:454 OpenZAP/1:4/ CHANNEL ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_ivr_originate.c:64 (OpenZAP/1:4/) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING going to sleep 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA going to sleep 2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5 2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 2 2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5 2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 1 2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:802 EVENT [ONHOOK][1:4] STATE [DIALING] 2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:838 Changing state on 1:4 from DIALING to DOWN 2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DOWN 2010-07-01 11:07:21.166199 [DEBUG] mod_openzap.c:1556 got FXO sig 1:4 [STOP] 2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2261 (OpenZAP/1:4/) Callstate Change RINGING -> HANGUP 2010-07-01 11:07:21.166199 [NOTICE] mod_openzap.c:1577 Hangup OpenZAP/1:4/ [CS_CONSUME_MEDIA] [NONE] 2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2277 Send signal OpenZAP/1:4/ [KILL] 2010-07-01 11:07:21.166199 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.166199 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_HANGUP 2010-07-01 11:07:21.166199 [DEBUG] zap_io.c:1388 channel done 1:4 2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:778 ANALOG CHANNEL 1:4 thread ended. 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP 2010-07-01 11:07:21.168191 [WARNING] mod_openzap.c:517 VETO Changing state on 1:4 from DOWN to HANGUP 2010-07-01 11:07:21.168191 [DEBUG] mod_openzap.c:556 OpenZAP/1:4/ CHANNEL HANGUP 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:46 OpenZAP/1:4/ Standard HANGUP, cause: NONE 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP going to sleep 2010-07-01 11:07:21.168191 [DEBUG] switch_ivr_originate.c:3369 Originate Resulted in Error Cause: 19 [NO_ANSWER] 2010-07-01 11:07:21.168191 [INFO] mod_dptools.c:2382 Originate Failed. Cause: NO_ANSWER 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/1:4/) State Change CS_HANGUP -> CS_REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:53 OpenZAP/1:4/ Standard REPORTING, cause: NONE 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING going to sleep 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/1:4/) State Change CS_REPORTING -> CS_DESTROY 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1175 Session 71 (OpenZAP/1:4/) Locked, Waiting on external entities 2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1193 Session 71 (OpenZAP/1:4/) Ended 2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1195 Close Channel OpenZAP/1:4/ [CS_DESTROY] But no ringing is heard. thanks, -:D On Thu, Jul 1, 2010 at 2:49 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: Default Dial Plan: action application bridge (Deya M) > > > ---------- Forwarded message ---------- > From: Deya M > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 1 Jul 2010 14:49:26 +1000 > Subject: Re: [Freeswitch-users] Default Dial Plan: action application > bridge > I generated samples, and started from scratch. > > I also made a copy of the conf after installation, rm it, and copied the > original conf before testing! > > -:D > > > On Thu, Jul 1, 2010 at 1:37 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: Query !! freeswitch, dingaling, jingle, skypopen >> (Anthony Minessale) >> 2. continue_on_fail and hangup_after_bridge with transfer >> (Vladimir Elizarov) >> 3. Re: SIP header on only one fork of a bridge (Tim St. Pierre) >> 4. Default Dial Plan: action application bridge (Deya M) >> 5. Re: Default Dial Plan: action application bridge >> (Anthony Minessale) >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 30 Jun 2010 17:40:27 -0500 >> Subject: Re: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, >> skypopen >> If he didn't like to joke, he'd never survive in telephony. >> >> >> On Wed, Jun 30, 2010 at 3:39 PM, Sameer Khan wrote: >> >>> no problem. but did't expect from the creator of mod_skypopen, the wonder >>> i would say >>> >>> >>> On Wed, Jun 30, 2010 at 9:19 PM, Giovanni Maruzzelli < >>> gmaruzz at celliax.org> wrote: >>> >>>> On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan >>>> wrote: >>>> > Mr Giovani what was that ? >>>> >>>> just jokin, nevermind >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.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 >>> >>> >> >> >> -- >> 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 >> >> >> ---------- Forwarded message ---------- >> From: Vladimir Elizarov >> To: freeswitch-users at lists.freeswitch.org >> Date: Thu, 1 Jul 2010 03:29:09 +0400 >> Subject: [Freeswitch-users] continue_on_fail and hangup_after_bridge with >> transfer >> I'm trying to make a dialplan: >> >> >> > >> expression="^(^([0-9]{10})$|^([0-9]{11})$|^([0-9]{12})$|^\+([0-9]{11})$)$"> >> > data="hangup_after_bridge=true"/> >> > application="set"data="continue_on_fail=NO_ROUTE_DESTINATION"/> >> >> >> >> > break="on-true"> >> > data="dtmf=WWWWWWWWWWWWWWWWWWWWW111222#WWWWWW89$2#@100"/> >> > >> data="sofia/gateway/gw1/89$2|sofia/gateway/gw2/89$2|sofia/gateway/gw3/89$2"/> >> >> >> >> >> >> >> > data="{execute_on_answer=send_dtmf\s${dtmf}}sofia/internal/${ >> real_dialed_number}@192.168.50.53:5061 >> "/> >> >> >> >> The logic of his work: if unavailable gateway to the next. if not >> available all the gateway to go to the extension transfer. >> A problem in If we get a code busy here, it is not satisfied >> hangup_after_bridge. The call goes to the transfer. Why is this so? >> >> trace: >> http://pastebin.freeswitch.org/13244 >> >> -- >> Best regards, Vladimir Elizarov >> >> >> >> >> ---------- Forwarded message ---------- >> From: "Tim St. Pierre" >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 30 Jun 2010 22:45:19 -0400 >> Subject: Re: [Freeswitch-users] SIP header on only one fork of a bridge >> Thanks all, >> >> I read that bit, but didn't fully understand it's implications. >> >> I'll give that a go. >> >> -Tim >> >> Steven Ayre wrote: >> > > > >> data="{this_is_global=true}[this_is_gw1_only=true]sofia/gateway/gw1/$1|[this_is_gw2_only=true]sofia/gateway/gw2/$1"/> >> > >> > >> > >> > On 28 June 2010 19:45, Tim St. Pierre > > > wrote: >> > >> > Hello list, >> > >> > I would like to bridge a call to multiple SIP endpoints, but add >> > different headers to each. >> > >> > I'm not entirely sure how to do this. I have no problem exporting a >> > SIP header that does what I >> > want for one destination, but I'm not sure how to set it for two. >> > >> > My application is that I want two IP phones to ring - one with the >> > internal ring-ring splash, the >> > others with a group-answer (single ring, then lamp flash only), for >> > administrative assistants, etc. >> > >> > How do I export different variables to each branch? >> > >> > Thanks! >> > >> > -Tim >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.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 >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Deya M >> To: freeswitch-users at lists.freeswitch.org >> Date: Thu, 1 Jul 2010 06:19:51 +0300 >> Subject: [Freeswitch-users] Default Dial Plan: action application bridge >> Hi, >> >> In the default dial plan, with two extensions defined, 1000 and 1001, when >> I call from 1000 to 1001, I always get Voicemail, using the default config >> files: conf/dialplan/default.xml >> >> I changed the following from : >> >> * * >> TO >> *> data="sofia/internal/${dialed_extension}%${domain_name}"/>* >> >> and it did work! Not sure if the first / default one, * > application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should >> work, but something is missing ? >> >> >> New: >> >> >> >> >> >> >> >> >> >> > data="transfer_ringback=$${hold_music}"/> >> >> >> >> >> >> > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >> var callgroup)}"/> >> >> > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> > data="sofia/internal/${dialed_extension}%${domain_name}"/> >> >> >> >> >> >> >> >> Old: >> >> >> >> >> >> >> >> >> >> >> > data="transfer_ringback=$${hold_music}"/> >> >> >> >> >> >> > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >> var callgroup)}"/> >> >> > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> >> >> >> >> >> >> >> :-D >> >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 30 Jun 2010 22:37:15 -0500 >> Subject: Re: [Freeswitch-users] Default Dial Plan: action application >> bridge >> you must have changed more than you think. >> >> you might want to diff you configs against the in-tree ones. >> >> >> On Wed, Jun 30, 2010 at 10:19 PM, Deya M wrote: >> >>> Hi, >>> >>> In the default dial plan, with two extensions defined, 1000 and 1001, >>> when I call from 1000 to 1001, I always get Voicemail, using the default >>> config files: conf/dialplan/default.xml >>> >>> I changed the following from : >>> >>> * * >>> TO >>> *>> data="sofia/internal/${dialed_extension}%${domain_name}"/>* >>> >>> and it did work! Not sure if the first / default one, * >> application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should >>> work, but something is missing ? >>> >>> >>> New: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> data="transfer_ringback=$${hold_music}"/> >>> >>> >>> >>> >>> >>> >> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >>> >> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >>> >> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >>> var callgroup)}"/> >>> >>> >> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >>> >> data="sofia/internal/${dialed_extension}%${domain_name}"/> >>> >>> >>> >>> >>> >>> >>> >>> Old: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> data="transfer_ringback=$${hold_music}"/> >>> >>> >>> >>> >>> >>> >> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >>> >> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >>> >> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >>> var callgroup)}"/> >>> >>> >> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >>> >>> >>> >>> >>> >>> >>> >>> :-D >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.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/20100701/20f37494/attachment-0001.html From irmatov at gmail.com Wed Jun 30 23:45:12 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Thu, 1 Jul 2010 11:45:12 +0500 Subject: [Freeswitch-users] mod_erlang_event problem In-Reply-To: References: <20100624145749.GA17555@hijacked.us> <20100625150622.GE17555@hijacked.us> Message-ID: On Wed, Jun 30, 2010 at 5:21 PM, Timur Irmatov wrote: > On Wed, Jun 30, 2010 at 12:02 PM, Timur Irmatov wrote: >> Ok, this is getting pretty confusing. > > It seems that my problem was related to host name part was not > properly configured in /etc/hosts. After fixing that *and* restarting > epmd, I was able to connect latest freeswitch and erlang. I was too quick to rejoice. It was really working for some time, then I have restarted freeswitch, and my problem returned. Could it be that there's some race condition between call initialisation/parking/whatever and mod_erlang_event interaction? Does anyone have any hints on debugging? What is the latest freeswitch and erlang that you people seem to have stable and working? -- Timur Irmatov, xmpp:irmatov at jabber.ru