From ribs at acm.org Wed Apr 1 00:00:53 2009 From: ribs at acm.org (Larry Edelstein) Date: Wed, 1 Apr 2009 00:00:53 -0700 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> Message-ID: <2021f8b20904010000i11611a1ewa4577d6d97ce9ba8@mail.gmail.com> You are then volunteering for something? 2009/3/31 > First off. I would not call it a "janitors project" since that may offend > some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would say > you have that backwards. There is nothing more effective in evolving > FreeSwitch than good documentation which helps further development and is an > important part of "customer service." Good customer service is then a part > of "sales and marketing." Much more often than not, It's sales and marketing > that is more important to making something a "real product" than > engineering. "Build it and they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org < > freeswitch-users at lists.freeswitch.org>; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's going to > keep growing as the word spreads. The core development team of Anthony, > Mike, and Brian are very appreciative of the community's help and > involvement in the project. Simply put: the community is awesome! > > Some have asked how they can help. Most of us are not software developers, > but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To > this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, takes > out the trash, replaces the toilet paper, etc. These are valuable services > that we sometimes take for granted. However, I think we can all appreciate > that the FreeSWITCH project would be better served if the developers could > focus on writing code, fixing bugs, etc. and not on the easier, > not-quite-as-important janitorial tasks. To that end we are inviting all who > wish to volunteer to please visit the above wiki page and check out some of > the projects listed so far. Email me off list if you'd like to volunteer to > help. I'm maintaining a list of "janitors" and what they are helping with. > If you have ideas for other janitor projects then by all means email them to > me and we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > New Low Prices on Dell Laptops - Starting at $399 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/6936e797/attachment.html From mszlazak at aol.com Wed Apr 1 00:12:33 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 03:12:33 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <2021f8b20904010000i11611a1ewa4577d6d97ce9ba8@mail.gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <2021f8b20904010000i11611a1ewa4577d6d97ce9ba8@mail.gmail.com> Message-ID: <8CB80B05D0C3E71-7D8-38C@webmail-mf17.sysops.aol.com> I just did, and it was suggestion. -----Original Message----- From: Larry Edelstein To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 12:00 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects You are then volunteering for something? 2009/3/31 First off. I would not call it a "janitors project" since that may offend some. A second problem is your notion that documentation is "not-quite-as-important" a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of "customer service." Good customer service is then a part of "sales and marketing." Much more often than not, It's sales and marketing that is more important to making something a "real product"? than engineering. "Build it and they will come" almost never works. Anyway, I think you need a new name for this project. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org ; freeswitch-dev at lists.freeswitch.org Sent: Tue, 31 Mar 2009 5:10 pm Subject: [Freeswitch-users] Call For Help: Janitor Projects Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a "janitor projects" wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say "janitor" projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of "janitors" and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009!? http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org New Low Prices on Dell Laptops - Starting at $399 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/9da061d6/attachment-0001.html From dujinfang at gmail.com Wed Apr 1 00:33:37 2009 From: dujinfang at gmail.com (seven) Date: Wed, 1 Apr 2009 15:33:37 +0800 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> Message-ID: <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> Agree, I think the author better to document the code first. For a simple example: if you add a new param or channel variable, at least should add an item to the wiki, so others knows there is a new variable and try that add add detailed explanation or experience further. On Apr 1, 2009, at 2:21 PM, mszlazak at aol.com wrote: > First off. I would not call it a "janitors project" since that may > offend some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many > would say you have that backwards. There is nothing more effective > in evolving FreeSwitch than good documentation which helps further > development and is an important part of "customer service." Good > customer service is then a part of "sales and marketing." Much more > often than not, It's sales and marketing that is more important to > making something a "real product" than engineering. "Build it and > they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org >; freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's > going to keep growing as the word spreads. The core development team > of Anthony, Mike, and Brian are very appreciative of the community's > help and involvement in the project. Simply put: the community is > awesome! > > Some have asked how they can help. Most of us are not software > developers, but that doesn't mean we can't help to grow the > FreeSWITCH ecosystem. To this end I've started a "janitor projects" > wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, > takes out the trash, replaces the toilet paper, etc. These are > valuable services that we sometimes take for granted. However, I > think we can all appreciate that the FreeSWITCH project would be > better served if the developers could focus on writing code, fixing > bugs, etc. and not on the easier, not-quite-as-important janitorial > tasks. To that end we are inviting all who wish to volunteer to > please visit the above wiki page and check out some of the projects > listed so far. Email me off list if you'd like to volunteer to help. > I'm maintaining a list of "janitors" and what they are helping with. > If you have ideas for other janitor projects then by all means email > them to me and we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > New Low Prices on Dell Laptops - Starting at $399 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/3a79ef33/attachment.html From raul at etellicom.com Wed Apr 1 01:29:56 2009 From: raul at etellicom.com (Raul Fragoso) Date: Wed, 01 Apr 2009 05:29:56 -0300 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> Message-ID: <1238574596.18630.64.camel@raul-laptop> Pardon my honesty, but I think you are the one who is getting this backwards. Firstly, I fail to see why a call for help with organizing and cleaning up the project documentation would offend someone by simply having "janitor" as the name. Have you ever heard the term "gatekeeper" before ? Would it offend you ? Think again. Secondly, FreeSWITCH is an open-source project, so forget the 'marketing & sales' crap in the context of documentation. The success of the project, which is growing incredibly fast, is built upon the collaboration of the community as a whole, and it's common sense that sharing the project tasks is a major necessary step to keep it going, just like a janitor is of primordial importance to keep an office building organized and clean. Last but not the least, I agree entirely with the fact that the core developers should be doing what they do it best, and that is, of course, development. I see this call for help request as an effective way of keeping them developing new features and improving the current functionality of FreeSWITCH while sharing the burden of documentation and organization. That's fair and sounds very logical to me. If you join the FreeSWITCH IRC channel and hang in there for a bit you will understand what I mean, most of the time these guys are busy responding to user questions or analyzing use cases that could be easily solved by checking a more organized documentation, and this is what Michael's request is all about. Regards, Raul On Wed, 2009-04-01 at 02:21 -0400, mszlazak at aol.com wrote: > First off. I would not call it a "janitors project" since that may > offend some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would > say you have that backwards. There is nothing more effective in > evolving FreeSwitch than good documentation which helps further > development and is an important part of "customer service." Good > customer service is then a part of "sales and marketing." Much more > often than not, It's sales and marketing that is more important to > making something a "real product" than engineering. "Build it and > they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > ; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's > going to keep growing as the word spreads. The core development team > of Anthony, Mike, and Brian are very appreciative of the community's > help and involvement in the project. Simply put: the community is > awesome! > > Some have asked how they can help. Most of us are not software > developers, but that doesn't mean we can't help to grow the FreeSWITCH > ecosystem. To this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, > takes out the trash, replaces the toilet paper, etc. These are > valuable services that we sometimes take for granted. However, I think > we can all appreciate that the FreeSWITCH project would be better > served if the developers could focus on writing code, fixing bugs, > etc. and not on the easier, not-quite-as-important janitorial tasks. > To that end we are inviting all who wish to volunteer to please visit > the above wiki page and check out some of the projects listed so far. > Email me off list if you'd like to volunteer to help. I'm maintaining > a list of "janitors" and what they are helping with. If you have ideas > for other janitor projects then by all means email them to me and > we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ______________________________________________________________________ > New Low Prices on Dell Laptops - Starting at $399 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 01:42:47 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 03:42:47 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> Message-ID: <1180B0C2-2791-4523-B0CD-1EB13213059F@freeswitch.org> What do you recommend calling it then? I wouldn't be offended by it ... and I can't think of any reason it would offend someone because it describes the task at hand. As far as documentation vs code... without the code there would be ZERO need for any documentation. The code is the hardest part to make sure it functions bug free. Developers are great at writing code but not the best at writing documentation, me included. It's the perfect place for anyone that wants to help out! I welcome anyone and everyone to the project in hopes that community members will help out! We have various IRC channels... #freeswitch, #freeswitch-dev, #freeswitch-docs and #freeswitch-social so join irc.freenode.net and get involved because you never know how it might change your life for the better! ;) /b Positive anything is better than negative thinking. On Apr 1, 2009, at 1:21 AM, mszlazak at aol.com wrote: > First off. I would not call it a "janitors project" since that may > offend some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many > would say you have that backwards. There is nothing more effective > in evolving FreeSwitch than good documentation which helps further > development and is an important part of "customer service." Good > customer service is then a part of "sales and marketing." Much more > often than not, It's sales and marketing that is more important to > making something a "real product" than engineering. "Build it and > they will come" almost never works. > > Anyway, I think you need a new name for this project. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/7d5dfda9/attachment.html From jason at jasonjgw.net Wed Apr 1 03:00:58 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Apr 2009 21:00:58 +1100 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> Message-ID: <20090401100058.GA15830@jdc.jasonjgw.net> seven wrote: > Agree, I think the author better to document the code first. Well, actually... it's already done. It's called API documentation, and consists of specially written comments in the code. This is not user-level documentation, however; it exists to help programmers who want to write applications or FreeSWITCH modules, or to participate in the development effort. Keep in mind also that this is a free software/open-source project; the developers are free to decide how best to spend their time. Personally, I would rather that they spend as much of the time as they wish writing and maintaining code. I've read enough of the code in FreeSWITCH to appreciate its high quality and the soundness of the design. It should also be remembered that the source code is the ultimate documentation, and everyone is free to look at it and to document (in their preferred natural language) what they find out. From dujinfang at gmail.com Wed Apr 1 03:45:28 2009 From: dujinfang at gmail.com (seven) Date: Wed, 1 Apr 2009 18:45:28 +0800 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <20090401100058.GA15830@jdc.jasonjgw.net> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> <20090401100058.GA15830@jdc.jasonjgw.net> Message-ID: <7FA73899-C2E5-4B69-B750-0E4700B8E26C@gmail.com> I know that. And I'd like to read code. Developers written great code and also plenty of comments(which is documentation) in code. However, there are sth. don't need to comment in code but should be available on wiki. E.g. I followed the svn commit log, and found sip_auth_username and sip_auth_password added, so I documented to the wiki. On Apr 1, 2009, at 6:00 PM, Jason White wrote: > seven wrote: >> Agree, I think the author better to document the code first. > > Well, actually... it's already done. It's called API documentation, > and > consists of specially written comments in the code. > > This is not user-level documentation, however; it exists to help > programmers > who want to write applications or FreeSWITCH modules, or to > participate in the > development effort. > > Keep in mind also that this is a free software/open-source project; > the > developers are free to decide how best to spend their time. I agree with you, whether of not document to wiki is up to the developers. But I just think it would be better(or more easier) if we(or others) can find all (including all the newest) params or features in wiki so we can try it and add document more on wiki. > > > Personally, I would rather that they spend as much of the time as > they wish > writing and maintaining code. > > I've read enough of the code in FreeSWITCH to appreciate its high > quality and > the soundness of the design. > > It should also be remembered that the source code is the ultimate > documentation, and everyone is free to look at it and to document > (in their > preferred natural language) what they find out. > > So do I. I'd like following the svn commit log to see what's new in there. But not all of us like to or have the time to read source code. Perhaps that's why we are here to help documenting.... > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Wed Apr 1 04:41:06 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 01 Apr 2009 13:41:06 +0200 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> Message-ID: <49D352D2.3070303@gmx.net> Hello Brian, I tried this (on trunk 12862), but still the same behaviour. It does not aks for a PIN. Neither when transfering directly to the conference nor by transfering to the dialplan extension where conference is handled. Best regards Peter Brian West schrieb: > Update again to svn trunk... btw 1.0.4 pre3 is out on > files.freeswitch.org > > /b > > On Mar 30, 2009, at 6:44 PM, Dan Le wrote: > >> I get similar behavior as Peter when trying to enter a locked >> conference. >> >> If I am just dialing from a phone to a conference (on a dialplan), it >> will properly lock me out. But if I do an originate command >> (originate sofia/internal/1001 &conference(3000)), it will drop me >> into the conference, even though it is suppose to be locked. >> >> I am using the released 1.0.3 tag. >> > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 lewisppp at gmail.com Wed Apr 1 03:41:44 2009 From: lewisppp at gmail.com (Lewis Liu) Date: Wed, 1 Apr 2009 18:41:44 +0800 Subject: [Freeswitch-users] Compiler error for Windows XP (SP2) Message-ID: <814e59990904010341y61b920c2h24bb8a50c8ae2f44@mail.gmail.com> We download FreeSWITCH from SVN Trunk and want to build it on MS Visual Studio 2008 with platform. But we got one error message when we build it. FreeSWITCH\libs\win32\sofia\debug\libsofia_sip_ua_static.lib is built fail. So many files are lost, such as mod_sofia.dll..... Could you help me me for this, Please?? Whether something is lost in MS Visual Studio 2008 ?? Thanks a lot!! Lewis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/b79932de/attachment.html From anthony.minessale at gmail.com Wed Apr 1 06:19:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 08:19:54 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> Message-ID: <191c3a030904010619i21ddd8fj81b020340907eb27@mail.gmail.com> have a look. http://www.google.com/search?q=janitor+project The phrase has already been coined. If you look closely we have 2 different perspectives in this thread. mszlazak is seeking more of the higher level user documentation, the holy grail magic documentation that is like the hitchhikers guide to the galaxy or harry potter's marauder's map can tune into what you need to know or what you don't understand and magically adjusts. This is normal, we have a lot of users like that. The majority of users will treat us like they are buying the software from us and impose their expectations on us. It's helpful to us, it lets us see things from their perspective. Seven is looking it at more from a developer's perspective, he's actually willing to take the time to add things to the wiki and he wants to understand how the code works. This is a good thing too, there are far less people of this type in our community but they are crucial. Core developers document by explaining what they are doing to people like Seven or by putting a reminder in the commit notes which are later translated into the CHANGELOG for the releases. Michael, the author of this thread has added countless pages of documentation to the wiki this way. It's easy to say the author should document everything. There is close to 300,000 lines of code in just the src directory in the FreeSWITCH tree (that is all code we wrote not counting any of the depends libs or any other form of pre-existing code). I personally wrote the majority of that code so, I really appricate it when the communiuty gives me a few minutes to take a break while they document it. The best people to document the high level fuctionality is not the author btw. It's the first few people who use it. Most likely they are developing a product from it and they intend to profit from it in one way or another and its a fair tradeoff to have the section of functionality explained to them in exchange from wikifying it from their perspective. The perspective of the author will be dry and mechinacal where that first-time-user version of the documentation will make much more sense to future readers. When it comes to the low level documentation, the C functions, we also need someone to help us with that if they feel there is not enough. We write code, we know how it works. If other people cannot figure out how it works, they will ask us and in the end it will be doucmented. About 5% or less of people in the community even have to look in the code for the core. The whole point of the FreeSWITCH design is to push everything up to scripts, remote connections and dialplan logic to let people concentrate on good ideas instead of the evil logic necessary to properly engineer a telephony engine. So I recommend anybody interested starts out making sure there is ample documentation for the embedded and external API for lua, js, perl, python, ESL etc. Then anybody who really likes C code can start with the module API layer and then dig deeper into the core code and learn how it works and if the documentation is not enough, add some, we appriciate any help we can get. 2009/4/1 > First off. I would not call it a "janitors project" since that may offend > some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would say > you have that backwards. There is nothing more effective in evolving > FreeSwitch than good documentation which helps further development and is an > important part of "customer service." Good customer service is then a part > of "sales and marketing." Much more often than not, It's sales and marketing > that is more important to making something a "real product" than > engineering. "Build it and they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org < > freeswitch-users at lists.freeswitch.org>; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's going to > keep growing as the word spreads. The core development team of Anthony, > Mike, and Brian are very appreciative of the community's help and > involvement in the project. Simply put: the community is awesome! > > Some have asked how they can help. Most of us are not software developers, > but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To > this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, takes > out the trash, replaces the toilet paper, etc. These are valuable services > that we sometimes take for granted. However, I think we can all appreciate > that the FreeSWITCH project would be better served if the developers could > focus on writing code, fixing bugs, etc. and not on the easier, > not-quite-as-important janitorial tasks. To that end we are inviting all who > wish to volunteer to please visit the above wiki page and check out some of > the projects listed so far. Email me off list if you'd like to volunteer to > help. I'm maintaining a list of "janitors" and what they are helping with. > If you have ideas for other janitor projects then by all means email them to > me and we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > New Low Prices on Dell Laptops - Starting at $399 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f8bc7440/attachment.html From mike at jerris.com Wed Apr 1 06:55:10 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 1 Apr 2009 09:55:10 -0400 Subject: [Freeswitch-users] Compiler error for Windows XP (SP2) In-Reply-To: <814e59990904010341y61b920c2h24bb8a50c8ae2f44@mail.gmail.com> References: <814e59990904010341y61b920c2h24bb8a50c8ae2f44@mail.gmail.com> Message-ID: <711C4390-ED0C-4A06-9AE8-652B24D0C776@jerris.com> If you try to build just the sofia library, what are the first few warnings and errors you get? Mike On Apr 1, 2009, at 6:41 AM, Lewis Liu wrote: > We download FreeSWITCH from SVN Trunk and want to build it on MS > Visual Studio 2008 with platform. > But we got one error message when we build it. > FreeSWITCH\libs\win32\sofia\debug\libsofia_sip_ua_static.lib is > built fail. > So many files are lost, such as mod_sofia.dll..... > Could you help me me for this, Please?? > Whether something is lost in MS Visual Studio 2008 ?? > Thanks a lot!! > Lewis From intralanman at freeswitch.org Wed Apr 1 06:59:15 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 01 Apr 2009 09:59:15 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <7FA73899-C2E5-4B69-B750-0E4700B8E26C@gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> <20090401100058.GA15830@jdc.jasonjgw.net> <7FA73899-C2E5-4B69-B750-0E4700B8E26C@gmail.com> Message-ID: <49D37333.5080701@freeswitch.org> seven wrote: > I know that. And I'd like to read code. Developers written great code > and also plenty of comments(which is documentation) in code. However, > there are sth. don't need to comment in code but should be available > on wiki. E.g. I followed the svn commit log, and found > sip_auth_username and sip_auth_password added, so I documented to the > wiki. > That's the right attitude to have... now if there were more people doing that and less people complaining like little school girls, we could actually reach the next level in Open-Sourcetopia. -Ray From anthony.minessale at gmail.com Wed Apr 1 07:30:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 09:30:06 -0500 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <49D352D2.3070303@gmx.net> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> <49D352D2.3070303@gmx.net> Message-ID: <191c3a030904010730h24946e50vf2ca7b2936f776d1@mail.gmail.com> pin checks and lock checks are both intentionally skipped on outbound calls transferred back to the conference. The idea is if you purposely placed an outbound call that was intended to land in the conference you would not want to do so only to tell them it's locked. I added a patch to trunk so you can override this with a variable originate {conference_enforce_security=true}sofia/internal/1001 &conference(3000) the same var can be used on inbound calls for the opposite effect On Wed, Apr 1, 2009 at 6:41 AM, Peter P GMX wrote: > Hello Brian, > > I tried this (on trunk 12862), but still the same behaviour. It does not > aks for a PIN. Neither when transfering directly to the conference nor > by transfering to the dialplan extension where conference is handled. > > Best regards > Peter > > > > Brian West schrieb: > > Update again to svn trunk... btw 1.0.4 pre3 is out on > > files.freeswitch.org > > > > /b > > > > On Mar 30, 2009, at 6:44 PM, Dan Le wrote: > > > >> I get similar behavior as Peter when trying to enter a locked > >> conference. > >> > >> If I am just dialing from a phone to a conference (on a dialplan), it > >> will properly lock me out. But if I do an originate command > >> (originate sofia/internal/1001 &conference(3000)), it will drop me > >> into the conference, even though it is suppose to be locked. > >> > >> I am using the released 1.0.3 tag. > >> > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/82d95c15/attachment.html From jmesquita at gmail.com Wed Apr 1 07:36:36 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 1 Apr 2009 11:36:36 -0300 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <49D37333.5080701@freeswitch.org> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> <20090401100058.GA15830@jdc.jasonjgw.net> <7FA73899-C2E5-4B69-B750-0E4700B8E26C@gmail.com> <49D37333.5080701@freeswitch.org> Message-ID: <2C42C47B-AD14-433F-B80F-9446E87D44F7@gmail.com> I am sorry, but I really have to comment this one. Why the fuck do we need to have sooo much politics on an open source project? Janitor, non-janitor, developer, non-developer, girl or boy, we are all trying to get this thing better, aren't we? So leave your fucking ego out of the question and get your ass doing something that will actually get this project somewhere like we all instead of trying to get yourself called something. You want the president title? Get it and start working. Tony is the master dude in this place because, like he said, he wrote most of the 300,000 line of code. That simple. The title "core developers team" (sounds great, doesn't it?) are because .... they do CORE! Wanna be called core developer, DO CORE! Anyway, my suggestion is, want something done? DO IT. Don't know how? Study! Don't want to know how ... buy Avaya or whatever. They will charge for your laziness. Sorry for the bad language. Mesquita On Apr 1, 2009, at 10:59 AM, Raymond Chandler wrote: > seven wrote: >> I know that. And I'd like to read code. Developers written great code >> and also plenty of comments(which is documentation) in code. However, >> there are sth. don't need to comment in code but should be available >> on wiki. E.g. I followed the svn commit log, and found >> sip_auth_username and sip_auth_password added, so I documented to the >> wiki. >> > That's the right attitude to have... now if there were more people > doing > that and less people complaining like little school girls, we could > actually reach the next level in Open-Sourcetopia. > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Apr 1 07:37:45 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 1 Apr 2009 07:37:45 -0700 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <49D37333.5080701@freeswitch.org> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> <20090401100058.GA15830@jdc.jasonjgw.net> <7FA73899-C2E5-4B69-B750-0E4700B8E26C@gmail.com> <49D37333.5080701@freeswitch.org> Message-ID: <37962F11-AAF0-42C5-97BA-C12A72194DA5@freeswitch.org> On Apr 1, 2009, at 6:59 AM, Raymond Chandler wrote: > seven wrote: >> I know that. And I'd like to read code. Developers written great code >> and also plenty of comments(which is documentation) in code. However, >> there are sth. don't need to comment in code but should be available >> on wiki. E.g. I followed the svn commit log, and found >> sip_auth_username and sip_auth_password added, so I documented to the >> wiki. >> > That's the right attitude to have... now if there were more people > doing > that and less people complaining like little school girls, we could > actually reach the next level in Open-Sourcetopia. > > -Ray First off, thank you all for your thoughts. This thread has yielded far more passion than I had hoped for. I consider that a good thing. It's okay for us to share differing opinions. Enthusiastic disagreements are better than ambivalence. :) Secondly, I just want to say that I like the term "janitor" because of its connotation. A janitor is someone who puts forth effort doing honorable work. A literal janitor is trusted with the keys to the office and leaves the workplace in a better condition than when he or she arrived. Another word for janitor is custodian. Please view the word in this positive light: a trusted worker whose contributions are valued by all. Thirdly, I want to thank people for stepping up. I've already received several private emails from volunteers. Please feel free to inundate my inbox! Lastly, I'd just like to thank Anthony, Brian, and Mike for devoting so much time and energy to FreeSWITCH. They've created a wonderful product, and they've also invested a lot of time answering my questions and those of others. I feel it's the least I can do to try and get that knowledge codified into a usable format so others can benefit also. Thanks again for your thoughts, ideas, and opinions. Keep them coming! We may just yet reach Open Sourcetopia. -MC From anthony.minessale at gmail.com Wed Apr 1 08:31:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 10:31:09 -0500 Subject: [Freeswitch-users] Another FreeSWITCH First! Message-ID: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> The FreeSWITCH team is excited to announce that FreeSWITCH is the first telephony application to support the new SIP 4.1 protocol specification. Unlike its predecessors, SIP 4.1 has been created with the collaboration of both the jabber foundation and the IETF. With this match made in heaven, one can now encapsulate an xml representation of a sip message, which in turn can encapsulate a standard SIP 2.0 message making it possible to do more than ever before. Other exciting features include: *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT with ease. *) Full circle presence: endpoints must subscribe to each character in the printable ASCII range that may be used to indicate presence and the server will send an xml notification to the client for each character that is enabled whenever a call takes place which in turn can be used to build a SIP 4.1 FYI packet that can be sent to all the neighboring SIP devices so they may send themselves a NOTIFY telling them that the light should blink if the same packet happens to be sent from a neighbor. Then when the neighbor wants to send a presence packet it establishes a dialog with the Third Party Presence Agent TPPA and leaves the message there. Then it sends the server a PRESENCE packet, which is then, relayed to the subscribers with the TPPA id so all the subscribers can connect to the TPPA server to make the little light blink. *) Retirement of SDP: SDP is deprecated in favor of a list of URL?s describing the desired codec. The UA can then request this URL and get the full details of the media requirements. The media port is negotiated through trial and error where the calling UA asks the called UA if the port it has guessed randomly is correct via direct TCP connection and an exchange of XML PORT MARKUP LANGUGE XPML INVITE bob at alice.com SIP 4.1 Content-type: sip-xml-encapsulated -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/d6ddf3b8/attachment.html From anthony.minessale at gmail.com Wed Apr 1 08:51:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 10:51:07 -0500 Subject: [Freeswitch-users] Long Lost Comments Surface, Now We Know... Message-ID: <191c3a030904010851m4c416ab8qfb41c04d731a8490@mail.gmail.com> In one of the most suprising events in current technology history in this modern era, the long lost comments to many of the now-adopted internet RFC's have finally surfaced. Aparently the mail server was misconfigured at "The Internet Society" and most of the comments were redirected to the local lost-and-found box on the server whey they sat for decades. In a suprising twist, it appears that RFC 2543, the predacessor to 3261 regarding the session initnation protocol had several critisisms that went unanswered. A few examples are included below. Some other comments in regards to RFC2833 (RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals) could not be published due to the graphic nature of the content. From: alice at anywhere.com to:comments at tis.org Subject: RFC 2543 Ahem, who gave you permission to use my name in your document? Also, how did you find out about me and Bob? Thanks to you it's all over the net >=0 From: jwlt at columbia.edu to:comments at tis.org Subject: RFC 2543 Are you guys sure about this? We were pretty drunk last night. I didn't think you would actually go through with it! lol I was just kiddding! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/57e729f8/attachment-0001.html From nik.middleton at noblesolutions.co.uk Wed Apr 1 09:07:36 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 1 Apr 2009 17:07:36 +0100 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> Message-ID: Well you almost had me there, but SIP over SMTP? That was too much. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 01 April 2009 16:31 To: Freeswitch-users Subject: [Freeswitch-users] Another FreeSWITCH First! The FreeSWITCH team is excited to announce that FreeSWITCH is the first telephony application to support the new SIP 4.1 protocol specification. Unlike its predecessors, SIP 4.1 has been created with the collaboration of both the jabber foundation and the IETF. With this match made in heaven, one can now encapsulate an xml representation of a sip message, which in turn can encapsulate a standard SIP 2.0 message making it possible to do more than ever before. Other exciting features include: *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT with ease. *) Full circle presence: endpoints must subscribe to each character in the printable ASCII range that may be used to indicate presence and the server will send an xml notification to the client for each character that is enabled whenever a call takes place which in turn can be used to build a SIP 4.1 FYI packet that can be sent to all the neighboring SIP devices so they may send themselves a NOTIFY telling them that the light should blink if the same packet happens to be sent from a neighbor. Then when the neighbor wants to send a presence packet it establishes a dialog with the Third Party Presence Agent TPPA and leaves the message there. Then it sends the server a PRESENCE packet, which is then, relayed to the subscribers with the TPPA id so all the subscribers can connect to the TPPA server to make the little light blink. *) Retirement of SDP: SDP is deprecated in favor of a list of URL's describing the desired codec. The UA can then request this URL and get the full details of the media requirements. The media port is negotiated through trial and error where the calling UA asks the called UA if the port it has guessed randomly is correct via direct TCP connection and an exchange of XML PORT MARKUP LANGUGE XPML INVITE bob at alice.com SIP 4.1 Content-type: sip-xml-encapsulated -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/8c815602/attachment.html From brian at freeswitch.org Wed Apr 1 09:15:24 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 11:15:24 -0500 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> Message-ID: You know you could write a transport plugin for Sofia that would do SIP over SMTP :P /b On Apr 1, 2009, at 11:07 AM, Nik Middleton wrote: > Well you almost had me there, but SIP over SMTP? That was too much. > > Regards, > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/ddfe7158/attachment.html From msc at freeswitch.org Wed Apr 1 09:34:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 09:34:54 -0700 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <191c3a030904010730h24946e50vf2ca7b2936f776d1@mail.gmail.com> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> <49D352D2.3070303@gmx.net> <191c3a030904010730h24946e50vf2ca7b2936f776d1@mail.gmail.com> Message-ID: <87f2f3b90904010934i567f9a15k13a33b2e125fd69c@mail.gmail.com> 2009/4/1 Anthony Minessale > pin checks and lock checks are both intentionally skipped on outbound calls > transferred back to the conference. > The idea is if you purposely placed an outbound call that was intended to > land in the conference > you would not want to do so only to tell them it's locked. > > I added a patch to trunk so you can override this with a variable > > originate {conference_enforce_security=true}sofia/internal/1001 > &conference(3000) > > the same var can be used on inbound calls for the opposite effect > > > > > > FYI this is now in the wiki: http://wiki.freeswitch.org/wiki/Channel_Variables#conference_enforce_security -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f2127449/attachment.html From msc at freeswitch.org Wed Apr 1 09:49:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 09:49:47 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre3 Now Available Message-ID: <87f2f3b90904010949x4d3fc2f8ia9feda2581956f68@mail.gmail.com> The FreeSWITCH team would like to let everyone know that the latest version is available. More information can be found here: http://www.freeswitch.org/node/172 By all means download, upgrade, test, and report back! Your feedback helps us make FreeSWITCH even better! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/deb90856/attachment.html From edpimentl at gmail.com Wed Apr 1 09:53:15 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 1 Apr 2009 12:53:15 -0400 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> Message-ID: <9dc4a1670904010953x4f1f6742h3fc1f355af23baa4@mail.gmail.com> LOVE!!!!! Now we can create Twitter-Voip apps.... Best regards, -E CEO and Founder Gpro.ws edpimentl [SKype | GoogleTalk | Twitter ] http://Twitter.com/edpimentl http://AskTwitR.com (Real Time Twitter Search & Reputation Management) http://TwiTR.Me (Cross Social Network Messaging Bus) http://TweetOnTV.net (Private Label Social TV Platform) http://TwebEX.com (Twitter Based Online Web Conference Platform) http://TwitrShare.com (Send Picture and Message to Tweet Contacts) http://Twookups.com (Twitter Matching Service) http://TweetUp.ws (Twitter based MeetUp service) 2009/4/1 Anthony Minessale > The FreeSWITCH team is excited to announce that FreeSWITCH is the first > telephony application to support the new SIP 4.1 protocol specification. > > Unlike its predecessors, SIP 4.1 has been created with the collaboration of > both the jabber foundation and the IETF. With this match made in heaven, > one can now encapsulate an xml representation of a sip message, which in > turn can encapsulate a standard SIP 2.0 message making it possible to do > more than ever before. > Other exciting features include: > > *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT with > ease. > > *) Full circle presence: endpoints must subscribe to each character in the > printable ASCII range that may be used to indicate presence and the server > will send an xml notification to the client for each character that is > enabled whenever a call takes place which in turn can be used to build a SIP > 4.1 FYI packet that can be sent to all the neighboring SIP devices so they > may send themselves a NOTIFY telling them that the light should blink if the > same packet happens to be sent from a neighbor. Then when the neighbor > wants to send a presence packet it establishes a dialog with the Third Party > Presence Agent TPPA and leaves the message there. Then it sends the server > a PRESENCE packet, which is then, relayed to the subscribers with the TPPA > id so all the subscribers can connect to the TPPA server to make the little > light blink. > > *) Retirement of SDP: SDP is deprecated in favor of a list of URL?s > describing the desired codec. The UA can then request this URL and get the > full details of the media requirements. The media port is negotiated > through trial and error where the calling UA asks the called UA if the port > it has guessed randomly is correct via direct TCP connection and an exchange > of XML PORT MARKUP LANGUGE XPML > > INVITE bob at alice.com SIP 4.1 > Content-type: sip-xml-encapsulated > > > > > To: bob at alice.com > From: alice at bob.com > Subject: SIP Rocks > ]]> > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/8f43290d/attachment-0001.html From brian at freeswitch.org Wed Apr 1 09:53:38 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 11:53:38 -0500 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre3 Now Available In-Reply-To: <87f2f3b90904010949x4d3fc2f8ia9feda2581956f68@mail.gmail.com> References: <87f2f3b90904010949x4d3fc2f8ia9feda2581956f68@mail.gmail.com> Message-ID: Which btw this is NOT an april fools joke! Its really 1.0.4 pre3 ;) /b On Apr 1, 2009, at 11:49 AM, Michael Collins wrote: > The FreeSWITCH team would like to let everyone know that the latest > version is available. More information can be found here: > http://www.freeswitch.org/node/172 > > By all means download, upgrade, test, and report back! Your feedback > helps us make FreeSWITCH even better! > -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 Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/23c6155a/attachment.html From mszlazak at aol.com Wed Apr 1 10:24:45 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 13:24:45 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <1238574596.18630.64.camel@raul-laptop> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> Message-ID: <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> Pardon me, but you speak only for yourself. I think Janitor is not an appropriate word. Second, 'marketing and sales' does not only mean making money. It also means 'selling' someone on the idea of trying something and effectively spreading the word. Third, the original developers can spend most of their time developing because they're the creators so they know very well what's going on with the code and don't need good documentation. Others need good documentation to effectively work with FS or do development. Currently the documentation is scattered, assumes to much and is outdated/incorrected. Also, there is a problem with not getting the "creators" involved with documentation since someone doing the documentation will have to ask them what's what. The "creators" never will be totally out of the loop nor should they be. This doesn't apply only here in this context but other similar ones as well. Keeping? "creators" from inteact with "customers" is one big reason so many start-ups fail. -----Original Message----- From: Raul Fragoso To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 1:29 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Pardon my honesty, but I think you are the one who is getting this backwards. Firstly, I fail to see why a call for help with organizing and cleaning up the project documentation would offend someone by simply having "janitor" as the name. Have you ever heard the term "gatekeeper" before ? Would it offend you ? Think again. Secondly, FreeSWITCH is an open-source project, so forget the 'marketing & sales' crap in the context of documentation. The success of the project, which is growing incredibly fast, is built upon the collaboration of the community as a whole, and it's common sense that sharing the project tasks is a major necessary step to keep it going, just like a janitor is of primordial importance to keep an office building organized and clean. Last but not the least, I agree entirely with the fact that the core developers should be doing what they do it best, and that is, of course, development. I see this call for help request as an effective way of keeping them developing new features and improving the current functionality of FreeSWITCH while sharing the burden of documentation and organization. That's fair and sounds very logical to me. If you join the FreeSWITCH IRC channel and hang in there for a bit you will understand what I mean, most of the time these guys are busy responding to user questions or analyzing use cases that could be easily solved by checking a more organized documentation, and this is what Michael's request is all about. Regards, Raul On Wed, 2009-04-01 at 02:21 -0400, mszlazak at aol.com wrote: > First off. I would not call it a "janitors project" since that may > offend some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would > say you have that backwards. There is nothing more effective in > evolving FreeSwitch than good documentation which helps further > development and is an important part of "customer service." Good > customer service is then a part of "sales and marketing." Much more > often than not, It's sales and marketing that is more important to > making something a "real product" than engineering. "Build it and > they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > ; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's > going to keep growing as the word spreads. The core development team > of Anthony, Mike, and Brian are very appreciative of the community's > help and involvement in the project. Simply put: the community is > awesome! > > Some have asked how they can help. Most of us are not software > developers, but that doesn't mean we can't help to grow the FreeSWITCH > ecosystem. To this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, > takes out the trash, replaces the toilet paper, etc. These are > valuable services that we sometimes take for granted. However, I think > we can all appreciate that the FreeSWITCH project would be better > served if the developers could focus on writing code, fixing bugs, > etc. and not on the easier, not-quite-as-important janitorial tasks. > To that end we are inviting all who wish to volunteer to please visit > the above wiki page and check out some of the projects listed so far. > Email me off list if you'd like to volunteer to help. I'm maintaining > a list of "janitors" and what they are helping with. If you have ideas > for other janitor projects then by all means email them to me and > we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ______________________________________________________________________ > New Low Prices on Dell Laptops - Starting at $399 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/325e0e6c/attachment.html From brian at freeswitch.org Wed Apr 1 10:39:44 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 12:39:44 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> Message-ID: <8534C3EE-DFE4-4743-8975-EAD9B2BCE1D8@freeswitch.org> Are you referring to PocketSphinx here? /b On Apr 1, 2009, at 12:24 PM, mszlazak at aol.com wrote: > Currently the documentation is scattered, assumes to much and is > outdated/incorrected. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/85fed388/attachment.html From mszlazak at aol.com Wed Apr 1 10:45:48 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 13:45:48 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <1180B0C2-2791-4523-B0CD-1EB13213059F@freeswitch.org> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1180B0C2-2791-4523-B0CD-1EB13213059F@freeswitch.org> Message-ID: <8CB8108D36771BE-458-31AD@webmail-dh09.sysops.aol.com> Call it what it is like "The Documentation Project" or something similar. Sure, if there was no code there is no FS but I didn't say the code is not important. I was taking a sales/marketing versus engineering analogy to this and only said that many would find it less important than good documentation if you are looking to get people to use FS and/or evolve the code. So as long as the creators of FS are willing to work to some extent on the documentation with a documentor, when one is needed, then this should work out. The creators have a very good understanding of FS which the documentor may not. On the other hand, the documentor doesn't have the creators background baggage which makes things seem obvious to the creator but isn't to users or even other developers. The creators and documentors working together will hopefully make the FS documentation accurate, not to presumptuous and easy to use. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 1:42 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects What do you recommend calling it then? ?I wouldn't be offended by it ... and I can't think of any reason it would offend someone because it describes the task at hand. ?As far as documentation vs code... without the code there would be ZERO need for any documentation. ?The code is the hardest part to make sure it functions bug free. ?Developers are great at writing code but not the best at writing documentation, me included. ?It's the perfect place for anyone that wants to help out! ?I welcome anyone and everyone to the project in hopes that community members will help out! ? We have various IRC channels... #freeswitch, #freeswitch-dev, #freeswitch-docs and #freeswitch-social so join irc.freenode.net and get involved because you never know how it might change your life for the better! ;) /b Positive anything is better than negative thinking. On Apr 1, 2009, at 1:21 AM, mszlazak at aol.com wrote: First off. I would not call it a "janitors project" since that may offend some. A second problem is your notion that documentation is "not-quite-as-important" a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of "customer service." Good customer service is then a part of "sales and marketing." Much more often than not, It's sales and marketing that is more important to making something a "real product"? than engineering. "Build it and they will come" almost never works. Anyway, I think you need a new name for this project. Brian West brian at freeswitch.org -- Meet us a ClueCon! ?http://www.cluecon.com = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/18d238f6/attachment-0001.html From brian at freeswitch.org Wed Apr 1 10:52:47 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 12:52:47 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB8108D36771BE-458-31AD@webmail-dh09.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1180B0C2-2791-4523-B0CD-1EB13213059F@freeswitch.org> <8CB8108D36771BE-458-31AD@webmail-dh09.sysops.aol.com> Message-ID: On Apr 1, 2009, at 12:45 PM, mszlazak at aol.com wrote: > Call it what it is like "The Documentation Project" or something > similar. Because its MORE than Documentation! So that name is silly! > > Sure, if there was no code there is no FS but I didn't say the code > is not important. I was taking a sales/marketing versus engineering > analogy to this and only said that many would find it less important > than good documentation if you are looking to get people to use FS > and/or evolve the code. So as long as the creators of FS are willing > to work to some extent on the documentation with a documentor, when > one is needed, then this should work out. The creators have a very > good understanding of FS which the documentor may not. On the other > hand, the documentor doesn't have the creators background baggage > which makes things seem obvious to the creator but isn't to users or > even other developers. The creators and documentors working together > will hopefully make the FS documentation accurate, not to > presumptuous and easy to use. Well if people join IRC... ask questions we do answer them... so if people don't understand something all they have to do is ask we won't bite. > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/8b735633/attachment.html From mszlazak at aol.com Wed Apr 1 10:56:02 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 13:56:02 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <191c3a030904010619i21ddd8fj81b020340907eb27@mail.gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <191c3a030904010619i21ddd8fj81b020340907eb27@mail.gmail.com> Message-ID: <8CB810A41678FCC-458-3259@webmail-dh09.sysops.aol.com> "The holy grail magic documentation that is like the hitchhikers guide to the galaxy or harry potter's marauder's map can tune into what you need to know or what you don't understand and magically adjusts."! Maybe your projecting or exaggerating but I didn't say anything like that. However, the important point was "we have a lot of users like that." Enough said. -----Original Message----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 6:19 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects have a look. http://www.google.com/search?q=janitor+project The phrase has already been coined. If you look closely we have 2 different perspectives in this thread. mszlazak is seeking more of the higher level user documentation, the holy grail magic documentation that is like the hitchhikers guide to the galaxy or harry potter's marauder's map can tune into what you need to know or what you don't understand and magically adjusts.? This is normal,?.? The majority of users will treat us like they are buying the software from us and impose their expectations on us.? It's helpful to us, it lets us see things from their perspective. Seven is looking it at more from a developer's perspective, he's actually willing to take the time to add things to the wiki and he wants to understand how the code works.? This is a good thing too, there are far less people of this type in our community but they are crucial.? Core developers document by explaining what they are doing to people like Seven or by putting a reminder in the commit notes which are later translated into the CHANGELOG for the releases.? Michael, the author of this thread has added countless pages of documentation to the wiki this way.? It's easy to say the author should document everything.? There is close to 300,000 lines of code in just the src directory in the FreeSWITCH tree (that is all code we wrote not counting any of the depends libs or any other form of pre-existing code).? I personally wrote the majority of that code so, I really appricate it when the communiuty gives me a few minutes to take a break while they document it.? The best people to document the high level fuctionality? is not the author btw.? It's the first few people who use it.? Most likely they are developing a product from it and they intend to profit from it in one way or another and its a fair tradeoff to have the section of functionality explained to them in exchange from wikifying it from their perspective.? The perspective of the author will be dry and mechinacal where that first-time-user version of the documentation will make much more sense to future readers.? When it comes to the low level documentation, the C functions, we also need someone to help us with that if they feel there is not enough.? We write code, we know how it works.? If other people cannot figure out how it works, they will ask us and in the end it will be doucmented.? About 5% or less of people in the community even have to look in the code for the core.? The whole point of the FreeSWITCH design is to push everything up to scripts, remote connections and dialplan logic to let people concentrate on good ideas instead of the evil logic necessary to properly engineer a telephony engine.? So I recommend anybody interested starts out making sure there is ample documentation for the embedded and external API for lua, js, perl, python, ESL etc.? Then anybody who really likes C code can start with the module API layer and then dig deeper into the core code and learn how it works and if the documentation is not enough, add some, we appriciate any help we can get. ? 2009/4/1 First off. I would not call it a "janitors project" since that may offend some. A second problem is your notion that documentation is "not-quite-as-important" a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of "customer service." Good customer service is then a part of "sales and marketing." Much more often than not, It's sales and marketing that is more important to making something a "real product"? than engineering. "Build it and they will come" almost never works. Anyway, I think you need a new name for this project. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org ; freeswitch-dev at lists.freeswitch.org Sent: Tue, 31 Mar 2009 5:10 pm Subject: [Freeswitch-users] Call For Help: Janitor Projects Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a "janitor projects" wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say "janitor" projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of "janitors" and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009!? http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org New Low Prices on Dell Laptops - Starting at $399 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/c9887216/attachment-0001.html From mszlazak at aol.com Wed Apr 1 10:56:37 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 13:56:37 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8534C3EE-DFE4-4743-8975-EAD9B2BCE1D8@freeswitch.org> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com><1238574596.18630.64.camel@raul-laptop><8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <8534C3EE-DFE4-4743-8975-EAD9B2BCE1D8@freeswitch.org> Message-ID: <8CB810A565E5D2A-458-3264@webmail-dh09.sysops.aol.com> nope -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 10:39 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Are you referring to PocketSphinx here?? /b On Apr 1, 2009, at 12:24 PM, mszlazak at aol.com wrote: ?Currently the documentation is scattered, assumes to much and is outdated/incorrected. Brian West brian at freeswitch.org -- Meet us a ClueCon! ?http://www.cluecon.com = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/8e0aee51/attachment.html From peter at cindyandpeter.com Wed Apr 1 10:58:57 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Wed, 1 Apr 2009 13:58:57 -0400 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> Message-ID: <02be01c9b2f3$87d77050$978650f0$@com> Excellent stuff Anthony! J SIP over SMTP could actually be useful in a push-to-talk type of scenario. Put the voice packets in an attachment. A slight delay, perhaps, but nicely encapsulated in a totally standard protocol. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: Wednesday, April 01, 2009 12:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Another FreeSWITCH First! Well you almost had me there, but SIP over SMTP? That was too much. Regards, _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 01 April 2009 16:31 To: Freeswitch-users Subject: [Freeswitch-users] Another FreeSWITCH First! The FreeSWITCH team is excited to announce that FreeSWITCH is the first telephony application to support the new SIP 4.1 protocol specification. Unlike its predecessors, SIP 4.1 has been created with the collaboration of both the jabber foundation and the IETF. With this match made in heaven, one can now encapsulate an xml representation of a sip message, which in turn can encapsulate a standard SIP 2.0 message making it possible to do more than ever before. Other exciting features include: *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT with ease. *) Full circle presence: endpoints must subscribe to each character in the printable ASCII range that may be used to indicate presence and the server will send an xml notification to the client for each character that is enabled whenever a call takes place which in turn can be used to build a SIP 4.1 FYI packet that can be sent to all the neighboring SIP devices so they may send themselves a NOTIFY telling them that the light should blink if the same packet happens to be sent from a neighbor. Then when the neighbor wants to send a presence packet it establishes a dialog with the Third Party Presence Agent TPPA and leaves the message there. Then it sends the server a PRESENCE packet, which is then, relayed to the subscribers with the TPPA id so all the subscribers can connect to the TPPA server to make the little light blink. *) Retirement of SDP: SDP is deprecated in favor of a list of URL's describing the desired codec. The UA can then request this URL and get the full details of the media requirements. The media port is negotiated through trial and error where the calling UA asks the called UA if the port it has guessed randomly is correct via direct TCP connection and an exchange of XML PORT MARKUP LANGUGE XPML INVITE bob at alice.com SIP 4.1 Content-type: sip-xml-encapsulated -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/ef4291f4/attachment.html From mszlazak at aol.com Wed Apr 1 11:02:22 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 14:02:22 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com><1180B0C2-2791-4523-B0CD-1EB13213059F@freeswitch.org><8CB8108D36771BE-458-31AD@webmail-dh09.sysops.aol.com> Message-ID: <8CB810B23ADBD5C-458-32D3@webmail-dh09.sysops.aol.com> Excellent! The core developers/creators should stay active in the documentation process. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 10:52 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects On Apr 1, 2009, at 12:45 PM, mszlazak at aol.com wrote: Call it what it is like "The Documentation Project" or something similar. Because its MORE than Documentation! ?So that name is silly! Sure, if there was no code there is no FS but I didn't say the code is not important.?I was taking a sales/marketing versus engineering analogy to this and?only said that many would find it less important than good documentation if you are looking to get people to use FS and/or evolve the code. So as long as the creators of FS are willing to work to some extent on the documentation with a documentor, when one is needed, then this should work out. The creators have a very good understanding of FS which the documentor may not. On the other hand, the documentor doesn't have the creators background baggage which makes things seem obvious to the creator but isn't to users or even other developers. The creators and documentors working together will hopefully make the FS documentation accurate, not to presumptuous and easy to use. Well if people join IRC... ask questions we do answer them... so if people don't understand something all they have to do is ask we won't bite. Brian West brian at freeswitch.org -- Meet us a ClueCon! ?http://www.cluecon.com = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/83b50667/attachment-0001.html From msc at freeswitch.org Wed Apr 1 11:18:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 11:18:26 -0700 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> Message-ID: <87f2f3b90904011118o28e4196bn50068353fa5ae8ea@mail.gmail.com> 2009/4/1 > Pardon me, but you speak only for yourself. I think Janitor is not an > appropriate word. > I *like* janitors. I *respect* janitors. They are *honorable* and * hard-working*. In short, we need janitors - people who are willing to roll up their sleeves and get work done. Let's agree to disagree on this word. If you don't like the word janitor then I will respect your viewpoint. Use the word "custodian" instead. However, the developers and all the core "power-users" have no qualms with the use of the word janitor. They will be called janitor projects; this point is not up for discussion. Let's all move on. As to your other points: yes, the core developers are involved in the documentation. They don't micromanage, but they give direction. When something is wrong they point it out. When there is a need, they make it known. When they get asked a lot of questions on a specific topic they tell me there's a need for documentation on the subject. Also, we have a number of users who are watching the mailing list and IRC channel who take it upon themselves to document the various nuggets of wisdom that get passed around in the threads. And I do my best to do same-day documentation when Anthony adds a new channel variable or new functionality to a module. As for documentation being outdated/scattered/incomplete/: Many of these observations are valid. There are serious needs - a lot of stuff needs cleaning up. (Which, ironically, is what *janitors* do very well.) However, let me make this point very clear: general statements like "the docs are out of date" are all but worthless. What we need are specific statements, like "I tried to follow the wiki instructions on pocketsphinx but I think they might be outdated or incorrect. May I discuss it with someone in the know?" All such specific comments are welcome. They can be sent to me personally, to this list, or on IRC. FYI, we do have a channel specifically for documentation discussion: #freeswitch-docs. Please join that channel to discuss this subject in real-time. All that being said, here's the bottom line: If you're willing to help then please do so. If you aren't sure where to start then contact me off list and we'll discuss it. If you have have positive feedback then please publish it publicly. If you have negative feedback, criticism, complaints, etc. then please send it to me in private. I've got my coveralls, my mop, and my bucket. Who's with me? :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/53b0ac30/attachment.html From raul at etellicom.com Wed Apr 1 11:21:40 2009 From: raul at etellicom.com (Raul Fragoso) Date: Wed, 01 Apr 2009 15:21:40 -0300 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: <02be01c9b2f3$87d77050$978650f0$@com> References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> <02be01c9b2f3$87d77050$978650f0$@com> Message-ID: <1238610100.10390.9.camel@raul-laptop> Agreed 100% ! That means we are all closer on taking 'mail-agents' to the holy-grail level of voice communications ! I wonder if SIP 4.1 UAS will also handle MX records ? That would be awesome ! I can't wait until we see something like mod_audio_spammer in FreeSWITCH, so those lovely marketing workers can give voice to their so much acclaimed phallic products. Regards, Raul On Wed, 2009-04-01 at 13:58 -0400, Peter J. Zandvoort wrote: > Excellent stuff Anthony! J > > > > SIP over SMTP could actually be useful in a push-to-talk type of > scenario. Put the voice packets in an attachment. A slight delay, > perhaps, but nicely encapsulated in a totally standard protocol. > > > > > > From:freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Nik Middleton > Sent: Wednesday, April 01, 2009 12:08 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Another FreeSWITCH First! > > > > > Well you almost had me there, but SIP over SMTP? That was too much. > > > > Regards, > > > > > ______________________________________________________________________ > From:freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony Minessale > Sent: 01 April 2009 16:31 > To: Freeswitch-users > Subject: [Freeswitch-users] Another FreeSWITCH First! > > > > > The FreeSWITCH team is excited to announce that FreeSWITCH is the > first telephony application to support the new SIP 4.1 protocol > specification. > > Unlike its predecessors, SIP 4.1 has been created with the > collaboration of both the jabber foundation and the IETF. With this > match made in heaven, one can now encapsulate an xml representation of > a sip message, which in turn can encapsulate a standard SIP 2.0 > message making it possible to do more than ever before. > Other exciting features include: > > *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT > with ease. > > *) Full circle presence: endpoints must subscribe to each character in > the printable ASCII range that may be used to indicate presence and > the server will send an xml notification to the client for each > character that is enabled whenever a call takes place which in turn > can be used to build a SIP 4.1 FYI packet that can be sent to all the > neighboring SIP devices so they may send themselves a NOTIFY telling > them that the light should blink if the same packet happens to be sent > from a neighbor. Then when the neighbor wants to send a presence > packet it establishes a dialog with the Third Party Presence Agent > TPPA and leaves the message there. Then it sends the server a > PRESENCE packet, which is then, relayed to the subscribers with the > TPPA id so all the subscribers can connect to the TPPA server to make > the little light blink. > > *) Retirement of SDP: SDP is deprecated in favor of a list of URL?s > describing the desired codec. The UA can then request this URL and > get the full details of the media requirements. The media port is > negotiated through trial and error where the calling UA asks the > called UA if the port it has guessed randomly is correct via direct > TCP connection and an exchange of XML PORT MARKUP LANGUGE XPML > > INVITE bob at alice.com SIP 4.1 > Content-type: sip-xml-encapsulated > > > > > To: bob at alice.com > From: alice at bob.com > Subject: SIP Rocks > ]]> > > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 11:23:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 13:23:48 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> Message-ID: <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> Did you follow the link I posted? http://www.google.com/search?q=janitor+project The linux kernel calls it the same thing and so do all the other project that come up in that search. Would you prefer "Custodial Engineering projects" I tried to be nice but you continue to perpetuate this thread. Another term you may not be familiar with is when someone who is outnumbered starts trying to get the last word on a mailing list or forum, they're called "trolls" Exactly how much have you contributed to this project other than complaints? You initially contacted us at our consulting address, where we then called you on the phone and helped you for 2 hours for free even though we know your goal is to develop a product from FreeSWITCH and most people in your position offer to pay us for our time. (make as many products as you want, that's why we made FreeSWITCH so good for you, but, usually if you want *that much* help you have to pay for it) You started using modules that were just written at the time you came around on a platform on which the module only was compiling for a week, give us a break..... We have all helped you on the list and documented things *for you* on several dozen occasions. I don't want anything in return but for you to please stop commenting on this thread. This is not a mob rule project, I will make the decisions for it when I see fit and when I seek the input of others, I ask for it and when I don't want any input I do whatever I want. It's a perk of running your own project. I personally don't care what Collins calls it, janitor project or whatever, at least he is show initiative and getting people involved. 2009/4/1 > Pardon me, but you speak only for yourself. I think Janitor is not an > appropriate word. > > Second, 'marketing and sales' does not only mean making money. It also > means 'selling' someone on the idea of trying something and effectively > spreading the word. > > Third, the original developers can spend most of their time developing > because they're the creators so they know very well what's going on with the > code and don't need good documentation. Others need good documentation to > effectively work with FS or do development. Currently the documentation is > scattered, assumes to much and is outdated/incorrected. Also, there is a > problem with not getting the "creators" involved with documentation since > someone doing the documentation will have to ask them what's what. The > "creators" never will be totally out of the loop nor should they be. This > doesn't apply only here in this context but other similar ones as well. > Keeping "creators" from inteact with "customers" is one big reason so many > start-ups fail. > > -----Original Message----- > From: Raul Fragoso > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, 1 Apr 2009 1:29 am > Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects > > Pardon my honesty, but I think you are the one who is getting this > > backwards. > > > Firstly, I fail to see why a call for help with organizing and cleaning > > up the project documentation would offend someone by simply having > > "janitor" as the name. Have you ever heard the term "gatekeeper" > > before ? Would it offend you ? Think again. > > > Secondly, FreeSWITCH is an open-source project, so forget the 'marketing > > & sales' crap in the context of documentation. The success of the > > project, which is growing incredibly fast, is built upon the > > collaboration of the community as a whole, and it's common sense that > > sharing the project tasks is a major necessary step to keep it going, > > just like a janitor is of primordial importance to keep an office > > building organized and clean. > > > Last but not the least, I agree entirely with the fact that the core > > developers should be doing what they do it best, and that is, of course, > > development. I see this call for help request as an effective way of > > keeping them developing new features and improving the current > > functionality of FreeSWITCH while sharing the burden of documentation > > and organization. That's fair and sounds very logical to me. If you join > > the FreeSWITCH IRC channel and hang in there for a bit you will > > understand what I mean, most of the time these guys are busy responding > > to user questions or analyzing use cases that could be easily solved by > > checking a more organized documentation, and this is what Michael's > > request is all about. > > > Regards, > > > Raul > > > On Wed, 2009-04-01 at 02:21 -0400, mszlazak at aol.com wrote: > > > First off. I would not call it a "janitors project" since that may > > > offend some. A second problem is your notion that documentation is > > > "not-quite-as-important" a task as writing code. I'm think many would > > > say you have that backwards. There is nothing more effective in > > > evolving FreeSwitch than good documentation which helps further > > > development and is an important part of "customer service." Good > > > customer service is then a part of "sales and marketing." Much more > > > often than not, It's sales and marketing that is more important to > > > making something a "real product" than engineering. "Build it and > > > they will come" almost never works. > > > > > > Anyway, I think you need a new name for this project. > > > > > > > > > > > > > > > > > > -----Original Message----- > > > From: Michael Collins > > > To: freeswitch-users at lists.freeswitch.org > > > ; > > > freeswitch-dev at lists.freeswitch.org > > > Sent: Tue, 31 Mar 2009 5:10 pm > > > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > > > > > Dear FreeSWITCH Community: > > > > > > As you know, FreeSWITCH has been growing leaps and bounds and it's > > > going to keep growing as the word spreads. The core development team > > > of Anthony, Mike, and Brian are very appreciative of the community's > > > help and involvement in the project. Simply put: the community is > > > awesome! > > > > > > Some have asked how they can help. Most of us are not software > > > developers, but that doesn't mean we can't help to grow the FreeSWITCH > > > ecosystem. To this end I've started a "janitor projects" wiki page: > > > > > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > > > > > We say "janitor" projects because they are things that help keep the > > > project clean and organized, just like the janitor cleans an office, > > > takes out the trash, replaces the toilet paper, etc. These are > > > valuable services that we sometimes take for granted. However, I think > > > we can all appreciate that the FreeSWITCH project would be better > > > served if the developers could focus on writing code, fixing bugs, > > > etc. and not on the easier, not-quite-as-important janitorial tasks. > > > To that end we are inviting all who wish to volunteer to please visit > > > the above wiki page and check out some of the projects listed so far. > > > Email me off list if you'd like to volunteer to help. I'm maintaining > > > a list of "janitors" and what they are helping with. If you have ideas > > > for other janitor projects then by all means email them to me and > > > we'll discuss them. > > > > > > Thanks again for being such a great community! > > > > > > -Michael S Collins > > > IRC: mercutioviz > > > > > > See you at ClueCon 2009! http://www.cluecon.com > > > > > > _______________________________________________ > > > > > > Freeswitch-users mailing list > > > > > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > > > > ______________________________________________________________________ > > > New Low Prices on Dell Laptops - Starting at $399 > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > New Low Prices on Dell Laptops - Starting at $399 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/3dfd737f/attachment-0001.html From grevenx at me.com Wed Apr 1 11:23:39 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Wed, 01 Apr 2009 20:23:39 +0200 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <87f2f3b90904010934i567f9a15k13a33b2e125fd69c@mail.gmail.com> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> <49D352D2.3070303@gmx.net> <191c3a030904010730h24946e50vf2ca7b2936f776d1@mail.gmail.com> <87f2f3b90904010934i567f9a15k13a33b2e125fd69c@mail.gmail.com> Message-ID: <95833A23-7543-4E88-8445-93FAE9CD00AE@me.com> You're one very fine janitor Michael! On the topic of the Janitor Project, this is how it should be. Devs give user feature => user documents new feature/behaviour. Even Andr? On 1. april. 2009, at 18.34, Michael Collins wrote: > > > 2009/4/1 Anthony Minessale > pin checks and lock checks are both intentionally skipped on > outbound calls transferred back to the conference. > The idea is if you purposely placed an outbound call that was > intended to land in the conference > you would not want to do so only to tell them it's locked. > > I added a patch to trunk so you can override this with a variable > > originate {conference_enforce_security=true}sofia/internal/1001 > &conference(3000) > > the same var can be used on inbound calls for the opposite effect > > > > > > > FYI this is now in the wiki: > http://wiki.freeswitch.org/wiki/Channel_Variables#conference_enforce_security > > -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/20090401/ef6a3779/attachment.html From intralanman at freeswitch.org Wed Apr 1 11:29:13 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 01 Apr 2009 14:29:13 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> Message-ID: <49D3B279.6040901@freeswitch.org> mszlazak at aol.com wrote: > Pardon me, but you speak only for yourself. I think Janitor is not an > appropriate word. you're welcome to your opinions, no matter how wrong they are > > Second, 'marketing and sales' does not only mean making money. It also > means 'selling' someone on the idea of trying something and > effectively spreading the word. > we don't try to sell anyone on the project... we'll tell you the pros and cons, you decide if the software meets your needs or not. > Third, the original developers can spend most of their time developing > because they're the creators so they know very well what's going on > with the code and don't need good documentation. Others need good > documentation to effectively work with FS or do development. Currently > the documentation is scattered, assumes to much and is > outdated/incorrected. maybe you could fix some of that since you seem to be very enlightened to its shortcomings? although, that might offend your delicate psyche since you'd basically be a "janitor" then. > Also, there is a problem with not getting the "creators" involved with > documentation since someone doing the documentation will have to ask > them what's what. The "creators" never will be totally out of the loop > nor should they be. This doesn't apply only here in this context but > other similar ones as well. Keeping "creators" from inteact with > "customers" is one big reason so many start-ups fail. > hmmm, maybe you're right... maybe the whole idea of hierarchy is entirely wrong. i guess we could expect tony to document his own code... while we're at it, let's suggest that microsoft has Bill Gates write documentation for windows and answer tech support calls, right? cus i mean, obviously everyone who writes code should obviously do everything else too, right? but i guess that doesn't work the other direction... cus if you don't know how to code, then you just can't code... its as simple as that. so now we have effectively halved (or better) the development activities of FreeSWITCH so there's less to document, but that's ok, because now there's plenty of people using it and not contributing anything back... and that's what open-source is really all about, right? btw, i'm just curious if you're an employee of a commercial entity that feels threatened by FreeSWITCH... what better way to decrease productivity than to split hairs over something so stupid as the name of an effort (janitor projects, in this case) that you're not going to take part in anyway. if i may ask, have you done anything constructive for the community at all? all i've seen of you from the mailing lists is non-constructive criticisms. not that we don't appreciate your trolling... its very entertaining to see how narrow-minded some people are. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/03b6bb8b/attachment.html From carlos.talbot at gmail.com Wed Apr 1 11:40:58 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 1 Apr 2009 13:40:58 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt Message-ID: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> Is there an interest in running FreeSWITCH on OpenWRT? I recently managed to compile and run a version for a MIPs based router, the Planex MZK-W04NU. This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP space. I was planning to setup a wiki page on compiling and configuring. regards, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/ddb2b8a5/attachment.html From msc at freeswitch.org Wed Apr 1 11:43:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 11:43:03 -0700 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <95833A23-7543-4E88-8445-93FAE9CD00AE@me.com> References: <49CBEA8D.4050901@gmx.net> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> <49D352D2.3070303@gmx.net> <191c3a030904010730h24946e50vf2ca7b2936f776d1@mail.gmail.com> <87f2f3b90904010934i567f9a15k13a33b2e125fd69c@mail.gmail.com> <95833A23-7543-4E88-8445-93FAE9CD00AE@me.com> Message-ID: <87f2f3b90904011143j465c462er2b44b173a2ba412e@mail.gmail.com> 2009/4/1 Even Andr? Fiskvik > You're one very fine janitor Michael! > How DARE you call me a janitor! :) > On the topic of the Janitor Project, this is how it should be. > Devs give user feature => user documents new feature/behaviour. > Thanks. This is totally reasonable. Power users and newbies both can add to the documentation. If anyone has questions about how to help or would like some pointers then by all means contact me off list. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/ea2a010f/attachment.html From timr at asteriasgi.com Wed Apr 1 11:46:39 2009 From: timr at asteriasgi.com (Tim Ringenbach) Date: Wed, 1 Apr 2009 13:46:39 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> Message-ID: <49D3B68F.7010806@asteriasgi.com> Anthony Minessale wrote: > Did you follow the link I posted? > http://www.google.com/search?q=janitor+project > > The linux kernel calls it the same thing and so do all the other > project that come up in that search. > > Would you prefer "Custodial Engineering projects" > It definitely is the commonly used term for that sort of thing. But I would tend to agree that I wouldn't expect people to get excited about volunteering to be a janitor. Any idea how successful those projects are at attracting volunteers? Sadly, I don't have a better suggestion. But no matter how much Michael says he loves janitors, to me a janitor is someone who has to clean up other people's crap (figuratively and sometimes literally). And I can see how that could fail to attract as many volunteers as the "Freeswitch Happy, Rich, and Well Endowed people" project might. > I tried to be nice but you continue to perpetuate this thread. > > Another term you may not be familiar with is when someone who is > outnumbered starts trying to > get the last word on a mailing list or forum, they're called "trolls" I always thought trolls had to be trying to really be considered a troll. Like if I were to post to this list trying to convince you all to give up on freeswitch and join the asterisk project, while knowing full well the history, and just trying to get a rise out of you. --Tim From msc at freeswitch.org Wed Apr 1 11:50:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 11:50:09 -0700 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> Message-ID: <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> 2009/4/1 Carlos Talbot > > Is there an interest in running FreeSWITCH on OpenWRT? I recently managed > to compile and run a version for a MIPs based router, the Planex MZK-W04NU. > This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support > (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks > online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP > space. > Just curious - is there a use case for doing this, other than the hobbyist who does it because it's cool? > > I was planning to setup a wiki page on compiling and configuring. Please do. We like to see all the different places and ways that people use FreeSWITCH. -MC > > > regards, > > Carlos > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/8503d3b2/attachment-0001.html From msc at freeswitch.org Wed Apr 1 11:54:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 11:54:43 -0700 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <49D3B68F.7010806@asteriasgi.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> <49D3B68F.7010806@asteriasgi.com> Message-ID: <87f2f3b90904011154l4c07279eg555c9574f168fb1a@mail.gmail.com> On Wed, Apr 1, 2009 at 11:46 AM, Tim Ringenbach wrote: > Anthony Minessale wrote: > > Did you follow the link I posted? > > http://www.google.com/search?q=janitor+project > > > > The linux kernel calls it the same thing and so do all the other > > project that come up in that search. > > > > Would you prefer "Custodial Engineering projects" > > > It definitely is the commonly used term for that sort of thing. But I > would tend to agree that I wouldn't expect people to get excited about > volunteering to be a janitor. Any idea how successful those projects are > at attracting volunteers? > > Sadly, I don't have a better suggestion. But no matter how much Michael > says he loves janitors, to me a janitor is someone who has to clean up > other people's crap (figuratively and sometimes literally). And I can > see how that could fail to attract as many volunteers as the "Freeswitch > Happy, Rich, and Well Endowed people" project might. Like I said: Grab a mop and bucket or get outta the way! It's time to take out the trash. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/acd46ee4/attachment.html From rupa at rupa.com Wed Apr 1 11:56:18 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 1 Apr 2009 13:56:18 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> Message-ID: 2009/4/1 Michael Collins > 2009/4/1 Carlos Talbot > >> >> Is there an interest in running FreeSWITCH on OpenWRT? I recently managed >> to compile and run a version for a MIPs based router, the Planex MZK-W04NU. >> This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support >> (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks >> online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP >> space. >> > > Just curious - is there a use case for doing this, other than the hobbyist > who does it because it's cool? > I could see using it as a standalone product for (very) small businesses or as a home gateway+phone. Guess the biggest issue would be lack of reasonable local storage for voicemail. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/347e0dd1/attachment.html From anthony.minessale at gmail.com Wed Apr 1 12:02:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 14:02:11 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <49D3B68F.7010806@asteriasgi.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> <49D3B68F.7010806@asteriasgi.com> Message-ID: <191c3a030904011202u74238814x78ee46c7674c994@mail.gmail.com> how about: "WALL-E projects" maybe Steve J will give us permission. On Wed, Apr 1, 2009 at 1:46 PM, Tim Ringenbach wrote: > Anthony Minessale wrote: > > Did you follow the link I posted? > > http://www.google.com/search?q=janitor+project > > > > The linux kernel calls it the same thing and so do all the other > > project that come up in that search. > > > > Would you prefer "Custodial Engineering projects" > > > It definitely is the commonly used term for that sort of thing. But I > would tend to agree that I wouldn't expect people to get excited about > volunteering to be a janitor. Any idea how successful those projects are > at attracting volunteers? > > Sadly, I don't have a better suggestion. But no matter how much Michael > says he loves janitors, to me a janitor is someone who has to clean up > other people's crap (figuratively and sometimes literally). And I can > see how that could fail to attract as many volunteers as the "Freeswitch > Happy, Rich, and Well Endowed people" project might. > > I tried to be nice but you continue to perpetuate this thread. > > > > Another term you may not be familiar with is when someone who is > > outnumbered starts trying to > > get the last word on a mailing list or forum, they're called "trolls" > I always thought trolls had to be trying to really be considered a > troll. Like if I were to post to this list trying to convince you all to > give up on freeswitch and join the asterisk project, while knowing full > well the history, and just trying to get a rise out of you. > > --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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/c5643291/attachment.html From carlos.talbot at gmail.com Wed Apr 1 12:02:06 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 1 Apr 2009 14:02:06 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> Message-ID: <5800526b0904011202s3f98757xc3df0327c2432ed1@mail.gmail.com> Until I figure out how much of a load it can handle for now it's just an experiment. :) I was motivated by two factors: - Kristin had ported FreeSWITCH to Astlinux, another uClib embedded environment. This sparked my interest in getting it to work on OpenWRT - Asterisk has been running on OpenWRT for a while so I wanted to see how difficult it would be to bring in FreeSWITCH. Carlos 2009/4/1 Michael Collins > 2009/4/1 Carlos Talbot > >> >> Is there an interest in running FreeSWITCH on OpenWRT? I recently managed >> to compile and run a version for a MIPs based router, the Planex MZK-W04NU. >> This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support >> (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks >> online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP >> space. >> > > Just curious - is there a use case for doing this, other than the hobbyist > who does it because it's cool? > > >> >> I was planning to setup a wiki page on compiling and configuring. > > > Please do. We like to see all the different places and ways that people use > FreeSWITCH. > > -MC > > >> >> >> regards, >> >> Carlos >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/b81c9443/attachment.html From stevecrozz at gmail.com Wed Apr 1 12:09:34 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 1 Apr 2009 12:09:34 -0700 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <5800526b0904011202s3f98757xc3df0327c2432ed1@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> <5800526b0904011202s3f98757xc3df0327c2432ed1@mail.gmail.com> Message-ID: <11990ade0904011209t133f12b4sc40246976be8972a@mail.gmail.com> Sounds like a fun project. I wouldn't worry too much about the lack of local storage space for voicemail. You can easily mount remote filesystems to increase storage capacity. I've done so using openwrt for my own projects using shfs, nfs, and next I want to try s3fs. --Stephen 2009/4/1 Carlos Talbot > Until I figure out how much of a load it can handle for now it's just an > experiment. :) > > I was motivated by two factors: > > - Kristin had ported FreeSWITCH to Astlinux, another uClib embedded > environment. This sparked my interest in getting it to work on OpenWRT > - Asterisk has been running on OpenWRT for a while so I wanted to see how > difficult it would be to bring in FreeSWITCH. > > Carlos > > 2009/4/1 Michael Collins > >> 2009/4/1 Carlos Talbot >> >> >>> Is there an interest in running FreeSWITCH on OpenWRT? I recently managed >>> to compile and run a version for a MIPs based router, the Planex MZK-W04NU. >>> This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support >>> (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks >>> online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP >>> space. >>> >> >> Just curious - is there a use case for doing this, other than the hobbyist >> who does it because it's cool? >> >> >>> >>> I was planning to setup a wiki page on compiling and configuring. >> >> >> Please do. We like to see all the different places and ways that people >> use FreeSWITCH. >> >> -MC >> >> >>> >>> >>> regards, >>> >>> Carlos >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f4b78d03/attachment-0001.html From cesar.bermudez at gmail.com Wed Apr 1 13:27:27 2009 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Wed, 1 Apr 2009 22:27:27 +0200 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> Message-ID: where can see and buy that router? 2009/4/1 Carlos Talbot > > Is there an interest in running FreeSWITCH on OpenWRT? I recently managed > to compile and run a version for a MIPs based router, the Planex MZK-W04NU. > This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support > (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks > online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP > space. > > I was planning to setup a wiki page on compiling and configuring. > > regards, > > Carlos > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f74c109c/attachment.html From mszlazak at aol.com Wed Apr 1 13:29:31 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 16:29:31 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com><1238574596.18630.64.camel@raul-laptop><8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> Message-ID: <8CB811FB2C391ED-698-1157@webmail-dd17.sysops.aol.com> You tried to be nice! Give me a break. Maybe try harder next time. -----Original Message----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 11:23 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Did you follow the link I posted? http://www.google.com/search?q=janitor+project The linux kernel calls it the same thing and so do all the other project that come up in that search. Would you prefer "Custodial Engineering projects" I tried to be nice but you continue to perpetuate this thread. Another term you may not be familiar with is when someone who is outnumbered starts trying to get the last word on a mailing list or forum, they're called "trolls" Exactly how much have you contributed to this project other than complaints? You initially contacted us at our consulting address, where we then called you on the phone and helped you for 2 hours for free even though we know your goal is to develop a product from FreeSWITCH and most people in your position offer to pay us for our time.? (make as many products as you want, that's why we made FreeSWITCH so good for you, but, usually if you want *that much* help you have to pay for it) You started using modules that were just written at the time you came around on a platform on which the module only was compiling for a week, give us a break..... We have all helped you on the list and documented things *for you* on several dozen occasions. I don't want anything in return but for you to please stop commenting on this thread. This is not a mob rule project, I will make the decisions for it when I see fit and when I seek the input of others, I ask for it and when I don't want any input I do whatever I want.? It's a perk of running your own project.? I personally don't care what Collins calls it, janitor project or whatever, at least he is show initiative and getting people involved. 2009/4/1 Pardon me, but you speak only for yourself. I think Janitor is not an appropriate word. Second, 'marketing and sales' does not only mean making money. It also means 'selling' someone on the idea of trying something and effectively spreading the word. Third, the original developers can spend most of their time developing because they're the creators so they know very well what's going on with the code and don't need good documentation. Others need good documentation to effectively work with FS or do development. Currently the documentation is scattered, assumes to much and is outdated/incorrected. Also, there is a problem with not getting the "creators" involved with documentation since someone doing the documentation will have to ask them what's what. The "creators" never will be totally out of the loop nor should they be. This doesn't apply only here in this context but other similar ones as well. Keeping? "creators" from inteact with "customers" is one big reason so many start-ups fail. -----Original Message----- From: Raul Fragoso To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 1:29 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Pardon my honesty, but I think you are the one who is getting this backwards. Firstly, I fail to see why a call for help with organizing and cleaning up the project documentation would offend someone by simply having "janitor" as the name. Have you ever heard the term "gatekeeper" before ? Would it offend you ? Think again. Secondly, FreeSWITCH is an open-source project, so forget the 'marketing & sales' crap in the context of documentation. The success of the project, which is growing incredibly fast, is built upon the collaboration of the community as a whole, and it's common sense that sharing the project tasks is a major necessary step to keep it going, just like a janitor is of primordial importance to keep an office building organized and clean. Last but not the least, I agree entirely with the fact that the core developers should be doing what they do it best, and that is, of course, development. I see this call for help request as an effective way of keeping them developing new features and improving the current functionality of FreeSWITCH while sharing the burden of documentation and organization. That's fair and sounds very logical to me. If you join the FreeSWITCH IRC channel and hang in there for a bit you will understand what I mean, most of the time these guys are busy responding to user questions or analyzing use cases that could be easily solved by checking a more organized documentation, and this is what Michael's request is all about. Regards, Raul On Wed, 2009-04-01 at 02:21 -0400, mszlazak at aol.com wrote: > First off. I would not call it a "janitors project" since that may > offend some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would > say you have that backwards. There is nothing more effective in > evolving FreeSwitch than good documentation which helps further > development and is an important part of "customer service." Good > customer service is then a part of "sales and marketing." Much more > often than not, It's sales and marketing that is more important to > making something a "real product" than engineering. "Build it and > they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > ; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's > going to keep growing as the word spreads. The core development team > of Anthony, Mike, and Brian are very appreciative of the community's > help and involvement in the project. Simply put: the community is > awesome! > > Some have asked how they can help. Most of us are not software > developers, but that doesn't mean we can't help to grow the FreeSWITCH > ecosystem. To this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, > takes out the trash, replaces the toilet paper, etc. These are > valuable services that we sometimes take for granted. However, I think > we can all appreciate that the FreeSWITCH project would be better > served if the developers could focus on writing code, fixing bugs, > etc. and not on the easier, not-quite-as-important janitorial tasks. > To that end we are inviting all who wish to volunteer to please visit > the above wiki page and check out some of the projects listed so far. > Email me off list if you'd like to volunteer to help. I'm maintaining > a list of "janitors" and what they are helping with. If you have ideas > for other janitor projects then by all means email them to me and > we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ______________________________________________________________________ > New Low Prices on Dell Laptops - Starting at $399 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org New Low Prices on Dell Laptops - Starting at $399 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/317c092a/attachment-0001.html From mszlazak at aol.com Wed Apr 1 13:31:01 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 16:31:01 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <49D3B279.6040901@freeswitch.org> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop><8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <49D3B279.6040901@freeswitch.org> Message-ID: <8CB811FE8046E7F-698-1171@webmail-dd17.sysops.aol.com> You missed the point again. But suffer fools to long. -----Original Message----- From: Raymond Chandler To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 11:29 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects mszlazak at aol.com wrote: Pardon me, but you speak only for yourself. I think Janitor is not an appropriate word. you're welcome to your opinions, no matter how wrong they are Second, 'marketing and sales' does not only mean making money. It also means 'selling' someone on the idea of trying something and effectively spreading the word. we don't try to sell anyone on the project... we'll tell you the pros and cons, you decide if the software meets your needs or not. Third, the original developers can spend most of their time developing because they're the creators so they know very well what's going on with the code and don't need good documentation. Others need good documentation to effectively work with FS or do development. Currently the documentation is scattered, assumes to much and is outdated/incorrected. maybe you could fix some of that since you seem to be very enlightened to its shortcomings? although, that might offend your delicate psyche since you'd basically be a "janitor" then. Also, there is a problem with not getting the "creators" involved with documentation since someone doing the documentation will have to ask them what's what. The "creators" never will be totally out of the loop nor should they be. This doesn't apply only here in this context but other similar ones as well. Keeping? "creators" from inteact with "customers" is one big reason so many start-ups fail. hmmm, maybe you're right... maybe the whole idea of hierarchy is entirely wrong. i guess we could expect tony to document his own code... while we're at it, let's suggest that microsoft has Bill Gates write documentation for windows and answer tech support calls, right? cus i mean, obviously everyone who writes code should obviously do everything else too, right? but i guess that doesn't work the other direction... cus if you don't know how to code, then you just can't code... its as simple as that. so now we have effectively halved (or better) the development activities of FreeSWITCH so there's less to document, but that's ok, because now there's plenty of people using it and not contributing anything back... and that's what open-source is really all about, right? btw, i'm just curious if you're an employee of a commercial entity that feels threatened by FreeSWITCH... what better way to decrease productivity than to split hairs over something so stupid as the name of an effort (janitor projects, in this case) that you're not going to take part in anyway. if i may ask, have you done anything constructive for the community at all? all i've seen of you from the mailing lists is non-constructive criticisms. not that we don't appreciate your trolling... its very entertaining to see how narrow-minded some people are. -Ray _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/4f8d39c9/attachment.html From valentin.doroga at pronexus.com Wed Apr 1 14:00:28 2009 From: valentin.doroga at pronexus.com (Valentin Doroga) Date: Wed, 1 Apr 2009 17:00:28 -0400 Subject: [Freeswitch-users] Nokia N800 In-Reply-To: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> Message-ID: <20090401210045.LGSN1685.tomts37-srv.bellnexxia.net@toip34-bus.srvr.bell.ca> There are some old binaries at: http://www.freeswitch.org/downloads/n800/ Is there a newer version? Any place with instruction to build? Val. From dave at 3c.co.uk Wed Apr 1 14:00:25 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 1 Apr 2009 14:00:25 -0700 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> Message-ID: Here's a sample SIP/SMTP INVITE (responses omitted for clarity) MAIL FROM: RCPT TO: DATA Call me . --Dave Sent from my iPhone On 1 Apr 2009, at 09:15, Brian West wrote: > You know you could write a transport plugin for Sofia that would do > SIP over SMTP :P > > /b > > On Apr 1, 2009, at 11:07 AM, Nik Middleton wrote: > >> Well you almost had me there, but SIP over SMTP? That was too much. >> >> Regards, >> > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f8fd5191/attachment.html From intralanman at freeswitch.org Wed Apr 1 14:04:36 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 01 Apr 2009 17:04:36 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB811FE8046E7F-698-1171@webmail-dd17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop><8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <49D3B279.6040901@freeswitch.org> <8CB811FE8046E7F-698-1171@webmail-dd17.sysops.aol.com> Message-ID: <49D3D6E4.1060301@freeswitch.org> mszlazak at aol.com wrote: > You missed the point again. But suffer fools to long. No, I think you missed the point... several times. The point that most of us are trying to make is "if you're not going to help, you have no room to talk". Although, I guess your approach works for you. If you're clearly outwitted, resort to name calling. I've seen a couple of people, including myself, ask if you've done anything except complain. I have not, however, seen you reply with anything intelligent or any contributions that you have made. So to try to make the point again. If you're not contributing anything, then leave us all alone. Hopefully, you're not so feeble-minded that you miss it twice in the same email. If you offer up ideas and they are accepted or considered, then you are a contributor. The point at which you offer your ideas and several members of the community, including the most involved, all disagree with you... you become a troll. It would be greatly apprciated by all persons involved if you, and your misguided opinions, would just concede and leave this thread alone. We now return you to the troll-free "Call For Help" -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/68f1613a/attachment.html From brian at freeswitch.org Wed Apr 1 14:11:13 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 16:11:13 -0500 Subject: [Freeswitch-users] Nokia N800 In-Reply-To: <20090401210045.LGSN1685.tomts37-srv.bellnexxia.net@toip34-bus.srvr.bell.ca> References: <20090401210045.LGSN1685.tomts37-srv.bellnexxia.net@toip34-bus.srvr.bell.ca> Message-ID: We haven't updated it recently... You should be able to use scratch box to accomplish it also. On that note please do not hijack threads... you clicked reply, changed the subject and body which causes it to thread your message with the original posters thread. So please in the future click new message and input freeswitch-users at lists.freeswitch.org Thanks, Brian On Apr 1, 2009, at 4:00 PM, Valentin Doroga wrote: > There are some old binaries at: > http://www.freeswitch.org/downloads/n800/ > > Is there a newer version? Any place with instruction to build? > Val. > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/44d85cd7/attachment-0001.html From stormin.normin at hotmail.co.uk Wed Apr 1 14:09:03 2009 From: stormin.normin at hotmail.co.uk (Stromin Normin) Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Message-ID: Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. _________________________________________________________________ Share your photos with Windows Live Photos ? Free. http://clk.atdmt.com/UKM/go/134665338/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/cabdfc34/attachment.html From toofics at gmail.com Wed Apr 1 13:19:26 2009 From: toofics at gmail.com (Victor Toofic) Date: Wed, 01 Apr 2009 14:19:26 -0600 Subject: [Freeswitch-users] Missing CHANNEL_HANGUP event in mod_event_socket Message-ID: <1238617166.3750.93.camel@ktulu> Hi all!! I'm stuck trying to use mod_event_socket in outbound mode. The problem that I'm facing is that while in a incoming call, using "myevents" to monitor for the channel's events.. the event CHANNEL_HANGUP sometimes arrives and sometimes doesn't. I can't figure it out why. The dialplan is: The process that handles the connection does: 1. connect 2. myevents (received: Reply-Text: +OK Events Enabled) 3. sendmsg\n call-command: execute\n execute-app-name: answer (received: Reply-Text: +OK) after this it waits for events and/or for the other party to hangup the call. (The DTMFs are for testing propourses). Sometimes the events that the process receives are: <<"CHANNEL_PARK">> <<"CHANNEL_EXECUTE">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"CHANNEL_EXECUTE">> <<"CHANNEL_ANSWER">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"DTMF">> <<"DTMF">> <<"CHANNEL_HANGUP">> (then it receives the "text/disconnect-notice" and the socket gets closed) and sometimes are: <<"CHANNEL_EXECUTE">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"CHANNEL_EXECUTE">> <<"CHANNEL_ANSWER">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"DTMF">> <<"DTMF">> (then it receives the "text/disconnect-notice" and the socket gets closed) As you can see, even sometimes the first CHANNEL_PARK event doesn't arrive. I'm very concerned about the missing CHANNEL_HANGUP event. In the other hand I was watching the events in a inbound connection to mod_event_socket with "event text all" and in this case there was no problem, all the events arrived as expected. Why in outbound mode some events get lost?? I'm missing something?? I've tried it in two different machines and the results are the same. I'm using FreeSWITCH Version 1.0.3 (exported) on linux. Thnks!! -- Regards.. Victor Toofic From rupa at rupa.com Wed Apr 1 14:13:04 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 1 Apr 2009 16:13:04 -0500 Subject: [Freeswitch-users] new module: mod_memcache Message-ID: Announcing a new module: mod_memcache Up until now one had two choices for storing arbitrary key/value pairs. hash or db. hash is fast, but it is local to the current FreeSWITCH instance. If you run multiple instances of FreeSWITCH then one could use db, an ODBC connection and a centralized database server (eg: postgresql). The choice was between fast but isolated or slow and distributed. memcached (http://www.danga.com/memcached/) is a high-performance, distributed memory object caching system, generic in nature, but intended for use in speeding up dynamic web applications by alleviating database load. Only now you can use it for dynamic phone applications. Check out the wiki page at: http://wiki.freeswitch.org/wiki/Mod_memcache Try this module out and file bug (jira) reports for problems / enhancement requests. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/beefa377/attachment.html From rupa at rupa.com Wed Apr 1 14:15:33 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 1 Apr 2009 16:15:33 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <11990ade0904011209t133f12b4sc40246976be8972a@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> <5800526b0904011202s3f98757xc3df0327c2432ed1@mail.gmail.com> <11990ade0904011209t133f12b4sc40246976be8972a@mail.gmail.com> Message-ID: s3fs would be ideal if this is a turnkey solution. still need local storage (flash) for the sqlite databases, but that shouldn't be very hard. 2009/4/1 Stephen Crosby > Sounds like a fun project. I wouldn't worry too much about the lack of > local storage space for voicemail. You can easily mount remote filesystems > to increase storage capacity. I've done so using openwrt for my own projects > using shfs, nfs, and next I want to try s3fs. > > --Stephen > > > 2009/4/1 Carlos Talbot > >> Until I figure out how much of a load it can handle for now it's just an >> experiment. :) >> >> I was motivated by two factors: >> >> - Kristin had ported FreeSWITCH to Astlinux, another uClib embedded >> environment. This sparked my interest in getting it to work on OpenWRT >> - Asterisk has been running on OpenWRT for a while so I wanted to see how >> difficult it would be to bring in FreeSWITCH. >> >> Carlos >> >> 2009/4/1 Michael Collins >> >>> 2009/4/1 Carlos Talbot >>> >>> >>>> Is there an interest in running FreeSWITCH on OpenWRT? I recently >>>> managed to compile and run a version for a MIPs based router, the Planex >>>> MZK-W04NU. This router has 32MB ram, 8MB flash, runs at 400MHz, draft N >>>> support (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 >>>> bucks online. I used a 2GB USB flash drive for the FreeSWITCH directory and >>>> SWAP space. >>>> >>> >>> Just curious - is there a use case for doing this, other than the >>> hobbyist who does it because it's cool? >>> >>> >>>> >>>> I was planning to setup a wiki page on compiling and configuring. >>> >>> >>> Please do. We like to see all the different places and ways that people >>> use FreeSWITCH. >>> >>> -MC >>> >>> >>>> >>>> >>>> regards, >>>> >>>> Carlos >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090401/88e399e4/attachment.html From anthony.minessale at gmail.com Wed Apr 1 14:18:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 16:18:03 -0500 Subject: [Freeswitch-users] Nokia N800 In-Reply-To: <20090401210045.LGSN1685.tomts37-srv.bellnexxia.net@toip34-bus.srvr.bell.ca> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> <20090401210045.LGSN1685.tomts37-srv.bellnexxia.net@toip34-bus.srvr.bell.ca> Message-ID: <191c3a030904011418x7d0b1e0fifb84e9c514dc51fd@mail.gmail.com> we relocated the machine with the build env for that, I'll try to find the time to resurrect it and make a new one. On Wed, Apr 1, 2009 at 4:00 PM, Valentin Doroga < valentin.doroga at pronexus.com> wrote: > There are some old binaries at: > http://www.freeswitch.org/downloads/n800/ > > Is there a newer version? Any place with instruction to build? > Val. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/c0b5cfc9/attachment.html From msc at freeswitch.org Wed Apr 1 14:18:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 14:18:44 -0700 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: <87f2f3b90904011418j371e1363m4ec1def2e9ba2818@mail.gmail.com> 2009/4/1 Stromin Normin > Hi, > > I've been asked to do some testing on Freeswitch by work, we currently use > Asterisk. I'm quite new to telephony so please go easy. > > I have FS setup on a windows box and at the moment I'm testing internal > calls only, when I transfer calls or call extensions everything sounds > great. The problem occurrs when I setup conferencing, people can log in ok > and we can talk, the trouble is as people start to talk a buzzing sound is > heard in the background, once the talking stops the buzzing stops. If the > person goes on mute there is no buzzing. > Out of curiosity, what kind of phones are you using? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/052816ec/attachment-0001.html From anthony.minessale at gmail.com Wed Apr 1 14:22:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 16:22:08 -0500 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: <191c3a030904011422r2f14d05ancae35f1ed3c8f09d@mail.gmail.com> the buzzing is probably a 60hz ground loop from the device that is calling in. Try using a different outlet, a different device, or if it's a cordless device like a laptop, try it with the power cable unplugged and only use battery to test it. Typically there is nothing we can do being on the receiving end of such noise. 2009/4/1 Stromin Normin > Hi, > > I've been asked to do some testing on Freeswitch by work, we currently use > Asterisk. I'm quite new to telephony so please go easy. > > I have FS setup on a windows box and at the moment I'm testing internal > calls only, when I transfer calls or call extensions everything sounds > great. The problem occurrs when I setup conferencing, people can log in ok > and we can talk, the trouble is as people start to talk a buzzing sound is > heard in the background, once the talking stops the buzzing stops. If the > person goes on mute there is no buzzing. > > Hopefully this is enough info cheers for any help. > > ------------------------------ > " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/b6734373/attachment.html From msc at freeswitch.org Wed Apr 1 14:22:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 14:22:16 -0700 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: References: Message-ID: <87f2f3b90904011422i32e2517dhd5cfc9414468c08@mail.gmail.com> Rupa, Thanks for adding to the project! Well done. -MC 2009/4/1 Rupa Schomaker > Announcing a new module: mod_memcache > > Up until now one had two choices for storing arbitrary key/value pairs. > hash or db. hash is fast, but it is local to the current FreeSWITCH > instance. If you run multiple instances of FreeSWITCH then one could use > db, an ODBC connection and a centralized database server (eg: postgresql). > > The choice was between fast but isolated or slow and distributed. > > memcached (http://www.danga.com/memcached/) is a high-performance, > distributed memory object caching system, generic in nature, but intended > for use in speeding up dynamic web applications by alleviating database > load. Only now you can use it for dynamic phone applications. > > Check out the wiki page at: http://wiki.freeswitch.org/wiki/Mod_memcache > > Try this module out and file bug (jira) reports for problems / enhancement > requests. > > -- > -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/20090401/22f090fe/attachment.html From anthony.minessale at gmail.com Wed Apr 1 14:23:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 16:23:48 -0500 Subject: [Freeswitch-users] Missing CHANNEL_HANGUP event in mod_event_socket In-Reply-To: <1238617166.3750.93.camel@ktulu> References: <1238617166.3750.93.camel@ktulu> Message-ID: <191c3a030904011423r29b5da21m4b5ca9d1fa2099ed@mail.gmail.com> its a race, sometimes the socket connection ends before the channel the linger socket command was added to tell FS to wait for the last channel event before ending the connection just send the command linger On Wed, Apr 1, 2009 at 3:19 PM, Victor Toofic wrote: > Hi all!! > > I'm stuck trying to use mod_event_socket in outbound mode. The problem > that I'm facing is that while in a incoming call, using "myevents" to > monitor for the channel's events.. the event CHANNEL_HANGUP sometimes > arrives and sometimes doesn't. I can't figure it out why. > > The dialplan is: > > > > > > > > > The process that handles the connection does: > > 1. connect > 2. myevents > (received: Reply-Text: +OK Events Enabled) > 3. sendmsg\n call-command: execute\n execute-app-name: answer > (received: Reply-Text: +OK) > > after this it waits for events and/or for the other party to hangup the > call. (The DTMFs are for testing propourses). > > Sometimes the events that the process receives are: > > <<"CHANNEL_PARK">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_ANSWER">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"DTMF">> > <<"DTMF">> > <<"CHANNEL_HANGUP">> > > (then it receives the "text/disconnect-notice" and the socket gets > closed) > > and sometimes are: > > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_ANSWER">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"DTMF">> > <<"DTMF">> > > (then it receives the "text/disconnect-notice" and the socket gets > closed) > > As you can see, even sometimes the first CHANNEL_PARK event doesn't > arrive. I'm very concerned about the missing CHANNEL_HANGUP event. > > In the other hand I was watching the events in a inbound connection to > mod_event_socket with "event text all" and in this case there was no > problem, all the events arrived as expected. > > Why in outbound mode some events get lost?? > I'm missing something?? > > I've tried it in two different machines and the results are the same. > I'm using FreeSWITCH Version 1.0.3 (exported) on linux. > > Thnks!! > > -- > Regards.. > Victor Toofic > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/59e29ec9/attachment.html From anthony.minessale at gmail.com Wed Apr 1 14:24:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 16:24:38 -0500 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: References: Message-ID: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> Thank you, You are brave to contribute something on April 1st =D I saw it go into tree everyone so it's real ;) 2009/4/1 Rupa Schomaker > Announcing a new module: mod_memcache > > Up until now one had two choices for storing arbitrary key/value pairs. > hash or db. hash is fast, but it is local to the current FreeSWITCH > instance. If you run multiple instances of FreeSWITCH then one could use > db, an ODBC connection and a centralized database server (eg: postgresql). > > The choice was between fast but isolated or slow and distributed. > > memcached (http://www.danga.com/memcached/) is a high-performance, > distributed memory object caching system, generic in nature, but intended > for use in speeding up dynamic web applications by alleviating database > load. Only now you can use it for dynamic phone applications. > > Check out the wiki page at: http://wiki.freeswitch.org/wiki/Mod_memcache > > Try this module out and file bug (jira) reports for problems / enhancement > requests. > > -- > -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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/d9f29280/attachment.html From rupa at rupa.com Wed Apr 1 14:31:09 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 1 Apr 2009 16:31:09 -0500 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> Message-ID: 2009/4/1 Anthony Minessale > Thank you, > > You are brave to contribute something on April 1st =D > I saw it go into tree everyone so it's real ;) > haha! I didn't even think of that. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/b00c6fad/attachment-0001.html From jmesquita at gmail.com Wed Apr 1 14:37:47 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 1 Apr 2009 18:37:47 -0300 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> Message-ID: <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> Congrats on the contribution Rupa. And thank you. Mesquita On Apr 1, 2009, at 6:31 PM, Rupa Schomaker wrote: > > > 2009/4/1 Anthony Minessale > Thank you, > > You are brave to contribute something on April 1st =D > I saw it go into tree everyone so it's real ;) > > haha! I didn't even think of that. > > -- > -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/20090401/fcfcf14f/attachment.html From stormin.normin at hotmail.co.uk Wed Apr 1 14:36:15 2009 From: stormin.normin at hotmail.co.uk (Stromin Normin) Date: Wed, 1 Apr 2009 22:36:15 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: Cheers for the replies. I'm not sure if I'm replying properly but here goes. I'm using Polycom 650 phones. I'm not really sure what a 60hz ground loop is so will need clarification, sorry I'm new to this. The phones are all on the same LAN and the conferencing is done on internal calls. Cheers From: stormin.normin at hotmail.co.uk To: freeswitch-users at lists.freeswitch.org Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now _________________________________________________________________ View your Twitter and Flickr updates from one place ? Learn more! http://clk.atdmt.com/UKM/go/137984870/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/88ba5c77/attachment.html From gmaruzz at celliax.org Wed Apr 1 15:10:35 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 2 Apr 2009 00:10:35 +0200 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: <7b197bef0904011510lb8adaadka391debfa2ec2a91@mail.gmail.com> To make a long story short, a ground loop is when an electric circuit is made between different audio device that are connected to the same electric power grid with badly grounded connections. This is an electrical problem generating noise, nothing to do with software. To test if this is the origin of your problem, try to use the devices unplugged from the electrical grid and check if the noise still there Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 2009/4/1 Stromin Normin : > Cheers for the replies.? I'm not sure if I'm replying properly but here > goes. > > I'm using Polycom 650 phones. > > I'm not really sure what a 60hz ground loop is so will need clarification, > sorry I'm new to this.? The phones are all on the same LAN and the > conferencing is done on internal calls. > > Cheers > > ________________________________ > From: stormin.normin at hotmail.co.uk > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 1 Apr 2009 22:09:03 +0100 > Subject: [Freeswitch-users] Buzzing when people speak in conference > > Hi, > > I've been asked to do some testing on Freeswitch by work, we currently use > Asterisk.? I'm quite new to telephony so please go easy. > > I have FS setup on a windows box and at the moment I'm testing internal > calls only, when I transfer calls or call extensions everything sounds > great.? The problem occurrs when I setup conferencing, people can log in ok > and we can talk, the trouble is as people start to talk a buzzing sound is > heard in the background, once the talking stops the buzzing stops.? If the > person goes on mute there is no buzzing. > > Hopefully this is enough info cheers for any help. > > ________________________________ > " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now > ________________________________ > Surfing the web just got more rewarding. Download the New Internet Explorer > 8 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Wed Apr 1 15:29:32 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 1 Apr 2009 17:29:32 -0500 Subject: [Freeswitch-users] mod_shout delay in trunk Message-ID: I've setup a conference bridge that has perpetual-sound set to a icecast stream. When the first person connects, there is an ~7s delay before any audio is heard. This is similar to a problem reported by Dan here and concluded with Tony adding the channel var "enable_file_write_buffering". The list discussion ended here: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011104.html I set this var in my dialplan: prior to joining the conference. The first person in still sees a 7s delay on audio the first time in. Like dan, I have icecast setup with burst_on_connect set to 1 but my burst_size is the default 64k so quite a bit of data. Has anyone been able to get an on-demand shoutcast stream from an icecast server to start immediately (or at least within a second)? Thanks. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/0faf0533/attachment.html From thorhs at basis.is Wed Apr 1 15:34:30 2009 From: thorhs at basis.is (Thorhallur Sverrisson) Date: Wed, 01 Apr 2009 22:34:30 +0000 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: <49D3EBF6.5050808@basis.is> The Polycom 650 is an IP phone, so the ground loop should not apply. Ground loops occur only in analog systems. As to what the buzzing is, I'm not sure. I have performed tests using Polycom 650s with out any sound artifacts. In fact the 650 audio has been flawless in my testing. Sorry I don't have a solution, just wanted to steer you away from a ground-loop debugging session. Thorhallur Stromin Normin wrote: > Cheers for the replies. I'm not sure if I'm replying properly but here > goes. > > I'm using Polycom 650 phones. > > I'm not really sure what a 60hz ground loop is so will need > clarification, sorry I'm new to this. The phones are all on the same > LAN and the conferencing is done on internal calls. > > Cheers > From msc at freeswitch.org Wed Apr 1 15:44:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 15:44:07 -0700 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: <49D3EBF6.5050808@basis.is> References: <49D3EBF6.5050808@basis.is> Message-ID: <87f2f3b90904011544t6d58c510ue707c98ac492bd2a@mail.gmail.com> That being the case, maybe a pcap of the audio might yield some clues? On Wed, Apr 1, 2009 at 3:34 PM, Thorhallur Sverrisson wrote: > The Polycom 650 is an IP phone, so the ground loop should not apply. > Ground loops occur only in analog systems. > > As to what the buzzing is, I'm not sure. I have performed tests using > Polycom 650s with out any sound artifacts. In fact the 650 audio has > been flawless in my testing. > > Sorry I don't have a solution, just wanted to steer you away from a > ground-loop debugging session. > > Thorhallur > > > Stromin Normin wrote: > > Cheers for the replies. I'm not sure if I'm replying properly but here > > goes. > > > > I'm using Polycom 650 phones. > > > > I'm not really sure what a 60hz ground loop is so will need > > clarification, sorry I'm new to this. The phones are all on the same > > LAN and the conferencing is done on internal calls. > > > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/78c9534f/attachment.html From hads at nice.net.nz Wed Apr 1 15:53:19 2009 From: hads at nice.net.nz (Hadley Rich) Date: Thu, 2 Apr 2009 11:53:19 +1300 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: <49D3EBF6.5050808@basis.is> References: <49D3EBF6.5050808@basis.is> Message-ID: <200904021153.19827.hads@nice.net.nz> On Thu, 02 Apr 2009 11:34:30 Thorhallur Sverrisson wrote: > The Polycom 650 is an IP phone, so the ground loop should not apply. > Ground loops occur only in analog systems. There is always an analog part to the system thus the potential for ground loops. It's common with snom phones when using a headset but I've not seen an issue with Polycom yet. hads -- http://nicegear.co.nz VoIP, DVB and other Linux compatible hardware. From elhodred at gmail.com Wed Apr 1 15:36:28 2009 From: elhodred at gmail.com (Alfonso Pinto) Date: Thu, 2 Apr 2009 00:36:28 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls Message-ID: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> Hi guys, I've using asterisk as PSTN gateway. When a call arrives from PSTN, I send the call to freeswitch and this route the call to a SIP gateway. When caller cancels the call (hangups before callee answers), I get this on asterisk CLI: chan_sip.c:13056 handle_response: Remote host can't match request CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 This is the sip call flow: u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 INVITE sip:666666666 at 1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "999999999" ;tag=as26208773. To: . Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Wed, 01 Apr 2009 21:03:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 29347 29347 IN IP4 2.2.2.2. s=session. c=IN IP4 2.2.2.2. t=0 0. m=audio 13846 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. From: "999999999" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Content-Length: 0. . U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. From: "999999999" ;tag=as26208773. To: ;tag=ceKFmNU84B90c. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Proxy-Authenticate: Digest realm="1.1.1.1", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, qop="auth". Content-Length: 0. . U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:666666666 at 1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "999999999" ;tag=as26208773. To: ;tag=ceKFmNU84B90c. Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 INVITE sip:666666666 at 1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. From: "999999999" ;tag=as26208773. To: . Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", algorithm=MD5, uri="sip:666666666 at 1.1.1.1", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, cnonce="47efcad4", nc=00000001. Date: Wed, 01 Apr 2009 21:03:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 29347 29348 IN IP4 2.2.2.2. s=session. c=IN IP4 2.2.2.2. t=0 0. m=audio 13846 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: "999999999" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Content-Length: 0. . U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 INVITE sip:666666666 at 3.3.3.3 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. Max-Forwards: 69. From: "999999999" ;tag=e050QBXFZXN6K. To: . Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 387. Remote-Party-ID: "999999999" ;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. s=FreeSWITCH. c=IN IP4 1.1.1.1. t=0 0. m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:3 GSM/8000. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:46:57 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 113193247 INVITE. Allow-Events: telephone-event. Content-Length: 0. . U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:46:57 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 113193247 INVITE. Allow-Events: telephone-event. Contact: . Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 300. . v=0. o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. s=SIP Call. c=IN IP4 3.3.3.3. t=0 0. m=audio 19398 RTP/AVP 18 13 101. c=IN IP4 3.3.3.3. a=rtpmap:18 G729/8000. a=rtpmap:13 CN/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:40. U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 292. . v=0. o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. s=FreeSWITCH. c=IN IP4 1.1.1.1. t=0 0. m=audio 20620 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. From: "999999999" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 CANCEL. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 481 Call/Transaction Does Not Exist. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 CANCEL. Content-Length: 0. . U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:46:57 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 113193247 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: . Content-Type: application/sdp. Content-Length: 300. . v=0. o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. s=SIP Call. c=IN IP4 3.3.3.3. t=0 0. m=audio 19398 RTP/AVP 18 13 101. c=IN IP4 3.3.3.3. a=rtpmap:18 G729/8000. a=rtpmap:13 CN/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:40. U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. Max-Forwards: 70. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 ACK. Contact: . Content-Length: 0. . U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 292. . v=0. o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. s=FreeSWITCH. c=IN IP4 1.1.1.1. t=0 0. m=audio 20620 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 104 BYE. User-Agent: Asterisk PBX. Max-Forwards: 70. Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, cnonce="6bc43301", nc=00000002. Content-Length: 0. . U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 104 BYE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Content-Length: 0. . U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. Max-Forwards: 70. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193248 BYE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. . U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:47:32 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. Content-Length: 0. CSeq: 113193248 BYE. . Please, can somebody tell me what is happening?. Thanks in advance. Regards. From chris at cloudtel.com Wed Apr 1 16:29:36 2009 From: chris at cloudtel.com (Chris Burns) Date: Wed, 1 Apr 2009 16:29:36 -0700 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: <7b197bef0904011510lb8adaadka391debfa2ec2a91@mail.gmail.com> References: <7b197bef0904011510lb8adaadka391debfa2ec2a91@mail.gmail.com> Message-ID: <200904011629.36433.chris@cloudtel.com> Try turning off comfort noise completely in the conference profile? My 650s sound great in conference w/ PCMU and G722 On April 1, 2009 03:10:35 pm Giovanni Maruzzelli wrote: > To make a long story short, a ground loop is when an electric circuit > is made between different audio device that are connected to the same > electric power grid with badly grounded connections. > > This is an electrical problem generating noise, nothing to do with > software. > > To test if this is the origin of your problem, try to use the devices > unplugged from the electrical grid and check if the noise still there > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > 2009/4/1 Stromin Normin : > > Cheers for the replies.? I'm not sure if I'm replying properly but here > > goes. > > > > I'm using Polycom 650 phones. > > > > I'm not really sure what a 60hz ground loop is so will need > > clarification, sorry I'm new to this.? The phones are all on the same LAN > > and the conferencing is done on internal calls. > > > > Cheers > > > > ________________________________ > > From: stormin.normin at hotmail.co.uk > > To: freeswitch-users at lists.freeswitch.org > > Date: Wed, 1 Apr 2009 22:09:03 +0100 > > Subject: [Freeswitch-users] Buzzing when people speak in conference > > > > Hi, > > > > I've been asked to do some testing on Freeswitch by work, we currently > > use Asterisk.? I'm quite new to telephony so please go easy. > > > > I have FS setup on a windows box and at the moment I'm testing internal > > calls only, when I transfer calls or call extensions everything sounds > > great.? The problem occurrs when I setup conferencing, people can log in > > ok and we can talk, the trouble is as people start to talk a buzzing > > sound is heard in the background, once the talking stops the buzzing > > stops.? If the person goes on mute there is no buzzing. > > > > Hopefully this is enough info cheers for any help. > > > > ________________________________ > > " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now > > ________________________________ > > Surfing the web just got more rewarding. Download the New Internet > > Explorer 8 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 16:34:51 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 18:34:51 -0500 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> Message-ID: I'm pretty sure this is a bug in Asterisk something to do with dialog matching... I think if you search the archives you'll see about it. /b On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666 at 1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666 at 3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" 999999999 at 3.3.3.3>;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: . > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: . > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: . > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > 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 Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/3d470bd8/attachment-0001.html From elhodred at gmail.com Wed Apr 1 16:41:57 2009 From: elhodred at gmail.com (Alfonso Pinto) Date: Thu, 2 Apr 2009 01:41:57 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> Message-ID: <8b3b7acc0904011641v2c7e3768ne72e06623ed064f2@mail.gmail.com> I've searched in google about it and only found a message about the same, Anthony asked for more information and nobody answer. I've tried with an IP phone (aastra 57i) and the same happens. Thank you 2009/4/2 Brian West : > I'm pretty sure this is a bug in Asterisk something to do with dialog > matching... I think if you search the archives you'll see about it. > /b > On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the ?call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666 at 1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666 at 3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" ;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" ;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: . > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: . > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: . > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > 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 > > Brian West > brian at freeswitch.org > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 16:46:35 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 18:46:35 -0500 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> Message-ID: <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> Follow this thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html /b On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666 at 1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666 at 3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" 999999999 at 3.3.3.3>;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: . > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: . > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: . > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > 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 Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/a9fd1495/attachment-0001.html From elhodred at gmail.com Wed Apr 1 17:09:42 2009 From: elhodred at gmail.com (Alfonso Pinto) Date: Thu, 2 Apr 2009 02:09:42 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> Message-ID: <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> One question more, maybe a stupid one: How can I search the archives? I didn't find nothing in lists.freeswitch.org. Regards 2009/4/2 Brian West : > Follow this > thread?http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html > /b > On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the ?call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666 at 1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666 at 3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" ;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" ;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: . > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: . > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: . > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > 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 > > Brian West > brian at freeswitch.org > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 17:19:06 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 19:19:06 -0500 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> Message-ID: If you go to google and input "site:lists.freeswitch.org blah" /b On Apr 1, 2009, at 7:09 PM, Alfonso Pinto wrote: > One question more, maybe a stupid one: How can I search the archives? > I didn't find nothing in lists.freeswitch.org. > > Regards Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/7e0d16cc/attachment.html From jason at jasonjgw.net Wed Apr 1 17:35:33 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 2 Apr 2009 11:35:33 +1100 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> Message-ID: <20090402003533.GA9849@jdc.jasonjgw.net> Alfonso Pinto wrote: > One question more, maybe a stupid one: How can I search the archives? http://www.gmane.org/ The searching tool they use, Xapian, tends to give good relevance ranking, at least in my experience. From sicfslist at gmail.com Wed Apr 1 17:56:32 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 1 Apr 2009 19:56:32 -0500 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: <200904011629.36433.chris@cloudtel.com> References: <7b197bef0904011510lb8adaadka391debfa2ec2a91@mail.gmail.com> <200904011629.36433.chris@cloudtel.com> Message-ID: <35b355e90904011756i3b3192fcm582b7e966e2397fb@mail.gmail.com> I have in a previous life seen this quite a bit with the PolyCom phones ... people tend to put their phone on the speaker on conference calls and I have seen this type of interference caused by a computer speaker and even a motorola cell phone. So I would first force everyone to use the handset 1st ... if that solves it then track down the guilty speaker or cell phone. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/9078884d/attachment.html From sicfslist at gmail.com Wed Apr 1 17:59:40 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 1 Apr 2009 19:59:40 -0500 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> Message-ID: <35b355e90904011759s43c91d4cg4d7a95b66d8027bd@mail.gmail.com> Rupa, This is a big contribution! Thanks! Can't wait to play with this. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/85c572d1/attachment-0001.html From toofics at gmail.com Wed Apr 1 17:44:13 2009 From: toofics at gmail.com (Victor Toofic) Date: Wed, 01 Apr 2009 18:44:13 -0600 Subject: [Freeswitch-users] Missing CHANNEL_HANGUP event in mod_event_socket In-Reply-To: <191c3a030904011423r29b5da21m4b5ca9d1fa2099ed@mail.gmail.com> References: <1238617166.3750.93.camel@ktulu> <191c3a030904011423r29b5da21m4b5ca9d1fa2099ed@mail.gmail.com> Message-ID: <1238633053.3750.99.camel@ktulu> thnks a lot!! I was getting scared.. lol Freeswitch rules!! On Wed, 2009-04-01 at 16:23 -0500, Anthony Minessale wrote: > its a race, > > sometimes the socket connection ends before the channel > > the linger socket command was added to tell FS to wait for the last > channel event before > ending the connection > > just send the command > > linger > > > > On Wed, Apr 1, 2009 at 3:19 PM, Victor Toofic > wrote: > Hi all!! > > I'm stuck trying to use mod_event_socket in outbound mode. The > problem > that I'm facing is that while in a incoming call, using > "myevents" to > monitor for the channel's events.. the event CHANNEL_HANGUP > sometimes > arrives and sometimes doesn't. I can't figure it out why. > > The dialplan is: > > > > > > > > > The process that handles the connection does: > > 1. connect > 2. myevents > (received: Reply-Text: +OK Events Enabled) > 3. sendmsg\n call-command: execute\n execute-app-name: answer > (received: Reply-Text: +OK) > > after this it waits for events and/or for the other party to > hangup the > call. (The DTMFs are for testing propourses). > > Sometimes the events that the process receives are: > > <<"CHANNEL_PARK">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_ANSWER">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"DTMF">> > <<"DTMF">> > <<"CHANNEL_HANGUP">> > > (then it receives the "text/disconnect-notice" and the socket > gets > closed) > > and sometimes are: > > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_ANSWER">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"DTMF">> > <<"DTMF">> > > (then it receives the "text/disconnect-notice" and the socket > gets > closed) > > As you can see, even sometimes the first CHANNEL_PARK event > doesn't > arrive. I'm very concerned about the missing CHANNEL_HANGUP > event. > > In the other hand I was watching the events in a inbound > connection to > mod_event_socket with "event text all" and in this case there > was no > problem, all the events arrived as expected. > > Why in outbound mode some events get lost?? > I'm missing something?? > > I've tried it in two different machines and the results are > the same. > I'm using FreeSWITCH Version 1.0.3 (exported) on linux. > > Thnks!! > > -- > Regards.. > Victor Toofic > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 18:06:19 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 20:06:19 -0500 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: <35b355e90904011759s43c91d4cg4d7a95b66d8027bd@mail.gmail.com> References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> <35b355e90904011759s43c91d4cg4d7a95b66d8027bd@mail.gmail.com> Message-ID: <7C5053E3-1F30-4D6C-B786-004931E813A1@freeswitch.org> At the very least you didn't say "I can't wait to play with it!" :P On Apr 1, 2009, at 7:59 PM, Shelby Ramsey wrote: > Rupa, > > This is a big contribution! Thanks! Can't wait to play with this. > > SDR Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/9cf25d0e/attachment.html From kristian.kielhofner at gmail.com Wed Apr 1 22:17:51 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 2 Apr 2009 01:17:51 -0400 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> Message-ID: <2d9149cd0904012217s692c4666tccb25b0db70b498b@mail.gmail.com> Carlos, I'm glad to see you've made some progress on your project. Keep us updated! 2009/4/1 Carlos Talbot : > > Is there an interest in running FreeSWITCH on OpenWRT? I recently managed to > compile and run a version for a MIPs based router, the Planex MZK-W04NU. > This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support > (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks > online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP > space. > > I was planning to setup a wiki page on compiling and configuring. > > regards, > > Carlos > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Wed Apr 1 23:09:56 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 2 Apr 2009 02:09:56 -0400 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <8b3b7acc0904011641v2c7e3768ne72e06623ed064f2@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <8b3b7acc0904011641v2c7e3768ne72e06623ed064f2@mail.gmail.com> Message-ID: <2d9149cd0904012309h5428ece5qce9a287b0972dd81@mail.gmail.com> I probably shouldn't be doing this for you, but... http://bugs.digium.com/view.php?id=14431 ;) On Wed, Apr 1, 2009 at 7:41 PM, Alfonso Pinto wrote: > I've searched in google about it and only found a message about the > same, Anthony asked for more information and nobody answer. > > I've tried with an IP phone (aastra 57i) and the same happens. > > Thank you > > 2009/4/2 Brian West : >> I'm pretty sure this is a bug in Asterisk something to do with dialog >> matching... I think if you search the archives you'll see about it. >> /b >> On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: >> >> Hi guys, >> >> I've using asterisk as PSTN gateway. When a call arrives from PSTN, I >> send the call to freeswitch and this route the call to a SIP gateway. >> >> When caller cancels the ?call (hangups before callee answers), I get >> this on asterisk CLI: >> >> chan_sip.c:13056 handle_response: Remote host can't match request >> CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. >> >> I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 >> >> This is the sip call flow: >> >> u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 >> INVITE sip:666666666 at 1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. >> From: "999999999" ;tag=as26208773. >> To: . >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 102 INVITE. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Date: Wed, 01 Apr 2009 21:03:12 GMT. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. >> Supported: replaces. >> Content-Type: application/sdp. >> Content-Length: 265. >> . >> v=0. >> o=root 29347 29347 IN IP4 2.2.2.2. >> s=session. >> c=IN IP4 2.2.2.2. >> t=0 0. >> m=audio 13846 RTP/AVP 18 101. >> a=rtpmap:18 G729/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=silenceSupp:off - - - -. >> a=ptime:20. >> a=sendrecv. >> >> >> U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 100 Trying. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 102 INVITE. >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 407 Proxy Authentication Required. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: ;tag=ceKFmNU84B90c. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 102 INVITE. >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Accept: application/sdp. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer. >> Proxy-Authenticate: Digest realm="1.1.1.1", >> nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, >> qop="auth". >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 >> ACK sip:666666666 at 1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. >> From: "999999999" ;tag=as26208773. >> To: ;tag=ceKFmNU84B90c. >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 102 ACK. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 >> INVITE sip:666666666 at 1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. >> From: "999999999" ;tag=as26208773. >> To: . >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 INVITE. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", >> algorithm=MD5, uri="sip:666666666 at 1.1.1.1", >> nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", >> response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, >> cnonce="47efcad4", nc=00000001. >> Date: Wed, 01 Apr 2009 21:03:12 GMT. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. >> Supported: replaces. >> Content-Type: application/sdp. >> Content-Length: 265. >> . >> v=0. >> o=root 29347 29348 IN IP4 2.2.2.2. >> s=session. >> c=IN IP4 2.2.2.2. >> t=0 0. >> m=audio 13846 RTP/AVP 18 101. >> a=rtpmap:18 G729/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=silenceSupp:off - - - -. >> a=ptime:20. >> a=sendrecv. >> >> >> U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 100 Trying. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 INVITE. >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 >> INVITE sip:666666666 at 3.3.3.3 SIP/2.0. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. >> Max-Forwards: 69. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: . >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> CSeq: 113193247 INVITE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer. >> Content-Type: application/sdp. >> Content-Disposition: session. >> Content-Length: 387. >> Remote-Party-ID: "999999999" ;screen=yes;privacy=off. >> . >> v=0. >> o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. >> s=FreeSWITCH. >> c=IN IP4 1.1.1.1. >> t=0 0. >> m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. >> a=rtpmap:18 G729/8000. >> a=rtpmap:4 G723/8000. >> a=rtpmap:3 GSM/8000. >> a=rtpmap:9 G722/8000. >> a=rtpmap:0 PCMU/8000. >> a=rtpmap:8 PCMA/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=rtpmap:13 CN/8000. >> a=ptime:20. >> >> >> U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 >> SIP/2.0 100 Trying. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Date: Fri, 05 Jan 2001 07:46:57 GMT. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> Server: Cisco-SIPGateway/IOS-12.x. >> CSeq: 113193247 INVITE. >> Allow-Events: telephone-event. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 >> SIP/2.0 183 Session Progress. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Date: Fri, 05 Jan 2001 07:46:57 GMT. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> Server: Cisco-SIPGateway/IOS-12.x. >> CSeq: 113193247 INVITE. >> Allow-Events: telephone-event. >> Contact: . >> Content-Disposition: session;handling=required. >> Content-Type: application/sdp. >> Content-Length: 300. >> . >> v=0. >> o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. >> s=SIP Call. >> c=IN IP4 3.3.3.3. >> t=0 0. >> m=audio 19398 RTP/AVP 18 13 101. >> c=IN IP4 3.3.3.3. >> a=rtpmap:18 G729/8000. >> a=rtpmap:13 CN/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:40. >> >> >> U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 183 Session Progress. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 INVITE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Accept: application/sdp. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer. >> Content-Type: application/sdp. >> Content-Disposition: session. >> Content-Length: 292. >> . >> v=0. >> o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. >> s=FreeSWITCH. >> c=IN IP4 1.1.1.1. >> t=0 0. >> m=audio 20620 RTP/AVP 18 101. >> a=rtpmap:18 G729/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=silenceSupp:off - - - -. >> a=ptime:20. >> >> >> U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 >> CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. >> From: "999999999" ;tag=as26208773. >> To: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 CANCEL. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 481 Call/Transaction Does Not Exist. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 CANCEL. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Date: Fri, 05 Jan 2001 07:46:57 GMT. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> Server: Cisco-SIPGateway/IOS-12.x. >> CSeq: 113193247 INVITE. >> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, >> SUBSCRIBE, NOTIFY, INFO. >> Allow-Events: telephone-event. >> Contact: . >> Content-Type: application/sdp. >> Content-Length: 300. >> . >> v=0. >> o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. >> s=SIP Call. >> c=IN IP4 3.3.3.3. >> t=0 0. >> m=audio 19398 RTP/AVP 18 13 101. >> c=IN IP4 3.3.3.3. >> a=rtpmap:18 G729/8000. >> a=rtpmap:13 CN/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:40. >> >> >> U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 >> ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. >> Max-Forwards: 70. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> CSeq: 113193247 ACK. >> Contact: . >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 INVITE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer. >> Content-Type: application/sdp. >> Content-Disposition: session. >> Content-Length: 292. >> . >> v=0. >> o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. >> s=FreeSWITCH. >> c=IN IP4 1.1.1.1. >> t=0 0. >> m=audio 20620 RTP/AVP 18 101. >> a=rtpmap:18 G729/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=silenceSupp:off - - - -. >> a=ptime:20. >> >> >> U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 >> ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 ACK. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 >> BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 104 BYE. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", >> algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", >> nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", >> response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, >> cnonce="6bc43301", nc=00000002. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 104 BYE. >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 >> BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. >> Max-Forwards: 70. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> CSeq: 113193248 BYE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Reason: Q.850;cause=16;text="NORMAL_CLEARING". >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Date: Fri, 05 Jan 2001 07:47:32 GMT. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> Server: Cisco-SIPGateway/IOS-12.x. >> Content-Length: 0. >> CSeq: 113193248 BYE. >> . >> >> Please, can somebody tell me what is happening?. >> >> Thanks in advance. >> >> 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 >> >> Brian West >> brian at freeswitch.org >> -- Meet us a ClueCon! ?http://www.cluecon.com >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From ashley.ohq at gmail.com Thu Apr 2 00:08:44 2009 From: ashley.ohq at gmail.com (Ashley van Gerven) Date: Thu, 2 Apr 2009 18:08:44 +1100 Subject: [Freeswitch-users] FS failover redundancy & load balancing Message-ID: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> Hi, I can't find much info on setting up a redundant or heavy load FreeSwitch implementation. Are there any links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? I imagine the entry level solution is to have two FS boxes configured identitcally, with redundant SBC software (recommendations?) in front, passing the calls to the primary FS box, or the backup FS box if the primary is not responding. Is that the easiest solution? What about a situation of having a level of concurrent calls beyond what one FS box can handle? I realise that would be a very large number of concurrent calls, but we would need a good plan on how to scale the systems. Are there recommendations for load balancing solutions? Either soft or hardware? My guess would be having 3 + 1 spare FS servers would work, where calls are distributed accross 3 FS boxes by a load balancer with one spare in event of failure. Also how would a FS box at max capacity behave? Does FS monitor available resources and reject the excess calls that it can't handle? Or would the load balancer have to be configured with the maximum number of calls per box? Would love to hear some experiences of deploying FS with failover & high load. Thanks Ash -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/807cb762/attachment.html From gmaruzz at celliax.org Thu Apr 2 01:35:21 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 2 Apr 2009 10:35:21 +0200 Subject: [Freeswitch-users] Skype interaction commands on skypiax Message-ID: <7b197bef0904020135j6b56662dy5a0dd2862ac4f35d@mail.gmail.com> Hi all, background: mod_skypiax is Skype compatible endpoint that allows incoming and outbound calls to/from the Skype network and SkypeOut service. It's seen by FS like other endpoints, so you can originate from sofia, bridge to skypiax, and connect the call to a landline number via SkypeOut service, for eg. skypiax endpoint use a Skype client to interact with the Skype network (see the wiki page for more details http://wiki.freeswitch.org/wiki/Skypiax). The news are: now you can send commands to the skype client related to a skyiax interface, both from the FS command line and programmatically (socket/API/esl/whatever) http://wiki.freeswitch.org/wiki/Skypiax#API_and_CLI_Commands This allow you to use directly the entire power of the Skype API ( https://developer.skype.com/Docs/ApiDoc ), for eg to send chat messages, interact with the buddy list, etc etc. Typing "console loglevel 9" at the FS command line allows you to see the Skype API answers from the Skype client instance. So, in short: you bring loglevel to 9 (so you can see the Skype API messages going back and forth), you use "sk" or "skypiax" to send Skype API commands to the Skype client instance. This way you can prototype extensions to the current mod_skypiax, that can then be implemented in C directly into the mod_skypiax source code. Please, let me know of extensions you would like to be integrated into the mod_skypiax code ;-). Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 From tristan at telemaque.fr Thu Apr 2 02:01:09 2009 From: tristan at telemaque.fr (Tristan) Date: Thu, 02 Apr 2009 11:01:09 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> Message-ID: <49D47ED5.5020409@telemaque.fr> Hi Ashley, One easy solution is to use a SIP proxy (opensips/kamailio/...) in front of FS boxes to load balance the charge between boxes. FS already has mechanisms to limit number of calls per boxes ( in switch.conf.xml: max-sessions and sessions-per-second ), that you can couple to load_balancing modules of the sip proxies. Of course you'll have to test to know how many session one box can handle, has it depends a lot on your usage of FS. Don' hesitate to join us on IRC if you want to discuss it ;) Regards, Gled Ashley van Gerven a ?crit : > Hi, > > I can't find much info on setting up a redundant or heavy load > FreeSwitch implementation. Are there any > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? > > I imagine the entry level solution is to have two FS boxes configured > identitcally, with > redundant SBC software (recommendations?) in front, passing the calls > to the primary FS box, > or the backup FS box if the primary is not responding. Is that the > easiest solution? > > What about a situation of having a level of concurrent calls beyond > what one FS box can handle? I realise > that would be a very large number of concurrent calls, but we would > need a good plan on how to scale the > systems. > > Are there recommendations for load balancing solutions? Either soft or > hardware? > > My guess would be having 3 + 1 spare FS servers would work, where > calls are distributed accross 3 FS boxes > by a load balancer with one spare in event of failure. > > Also how would a FS box at max capacity behave? Does FS monitor > available resources and reject the > excess calls that it can't handle? Or would the load balancer have to > be configured with the maximum number > of calls per box? > > Would love to hear some experiences of deploying FS with failover & > high load. > > > Thanks > Ash > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/c2aa1062/attachment.html From sridhart at alcatel-lucent.com Thu Apr 2 01:58:30 2009 From: sridhart at alcatel-lucent.com (Rajagopal, Sridhar (Sridhar)) Date: Thu, 2 Apr 2009 14:28:30 +0530 Subject: [Freeswitch-users] Dialplan for OPTIONS packet Message-ID: <9389DD3DDD6B9144B147CE564C6599B902D22BA2C9@INBANSXCHMBSA3.in.alcatel-lucent.com> Hi all, Whenever freeswitch recieves INVITE SIP packet, It forwards the packet based on the dial plan. I want to use the same dial plan to forward incoming OPTIONS packet. Please let me know If I need to write my own code for that or is there any such option in our code base. Regards, Sridhar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/d297f341/attachment.html From solko at gcdf.pl Thu Apr 2 02:13:48 2009 From: solko at gcdf.pl (Szymon Olko) Date: Thu, 02 Apr 2009 11:13:48 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> Message-ID: <49D481CC.70102@gcdf.pl> Ashley van Gerven pisze: > Hi, > > I can't find much info on setting up a redundant or heavy load > FreeSwitch implementation. Are there any > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? > > I imagine the entry level solution is to have two FS boxes configured > identitcally, with > redundant SBC software (recommendations?) in front, passing the calls to > the primary FS box, > or the backup FS box if the primary is not responding. Is that the > easiest solution? > > What about a situation of having a level of concurrent calls beyond what > one FS box can handle? I realise > that would be a very large number of concurrent calls, but we would need > a good plan on how to scale the > systems. > > Are there recommendations for load balancing solutions? Either soft or > hardware? > > My guess would be having 3 + 1 spare FS servers would work, where calls > are distributed accross 3 FS boxes > by a load balancer with one spare in event of failure. > > Also how would a FS box at max capacity behave? Does FS monitor > available resources and reject the > excess calls that it can't handle? Or would the load balancer have to be > configured with the maximum number > of calls per box? I did not think yet about HA nor LB. I tested how FS handles high load. All my calls are placed in mod_conference. When cpu usage gets it's limits then new calls can be placed but sound quality is getting worst with every next call. When calls are hanged up then sound gets better. I did not test it to see what happens when more and more calls are created. FS has very low memory consumption and I think that CPU is the limit. I did not notice any monitoring of CPU usage by FS, but my installation is limited to only few modules, so maybe I'm missing something. > > Would love to hear some experiences of deploying FS with failover & high My failover is currently made by shell script which every 10 seconds check for working FS and restarts it if it does not work. I use svn trunk so crash happens once a while, but they are successfully fixed by developers. Once there was a problem that conference module was stuck and did not respond to my commands. I made script with netcat which checks once a while for response and restarts if there is none. > load. > > > Thanks > Ash > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 2 04:55:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 06:55:02 -0500 Subject: [Freeswitch-users] Dialplan for OPTIONS packet In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B902D22BA2C9@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B902D22BA2C9@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <43B08789-DAE7-461A-BA73-3C73B9EAB7DC@freeswitch.org> Can you describe the reasoning behind needing to route option packets via the dialplan? /b On Apr 2, 2009, at 3:58 AM, Rajagopal, Sridhar (Sridhar) wrote: > Hi all, > > Whenever freeswitch recieves INVITE SIP packet, It forwards the > packet based on the dial plan. I want to use the same dial plan to > forward incoming OPTIONS packet. Please let me know If I need to > write my own code for that or is there any such option in our code > base. > > Regards, > Sridhar > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/d2e96f28/attachment.html From bmsword at gmail.com Thu Apr 2 00:29:14 2009 From: bmsword at gmail.com (bmsword) Date: Thu, 2 Apr 2009 15:29:14 +0800 Subject: [Freeswitch-users] about freeswitch conference References: <200904021524116567464@gmail.com> Message-ID: <200904021529137966712@gmail.com> hi,all I want to use another softswitch conference that has been deployed in freeswitch,How should I do? andy 2009-04-02 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/191beb1a/attachment-0001.html From stormin.normin at hotmail.co.uk Thu Apr 2 02:20:25 2009 From: stormin.normin at hotmail.co.uk (Stromin Normin) Date: Thu, 2 Apr 2009 10:20:25 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: Thanks for taking the time to help me. Giovanni, I assume you turn comfort noise off by setting it to 0 which I've now done. How can I tell which codecs I'm using in conference and how would I change them. The sound is ok on everything else. Thanks again From: stormin.normin at hotmail.co.uk To: freeswitch-users at lists.freeswitch.org Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now _________________________________________________________________ View your Twitter and Flickr updates from one place ? Learn more! http://clk.atdmt.com/UKM/go/137984870/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/c3db81ee/attachment.html From bmsword at gmail.com Thu Apr 2 02:26:29 2009 From: bmsword at gmail.com (bmsword) Date: Thu, 2 Apr 2009 17:26:29 +0800 Subject: [Freeswitch-users] about freeswitch conference References: <200904021524116567464@gmail.com> Message-ID: <200904021726283757151@gmail.com> hi,all I want to use another softswitch conference that has been deployed in freeswitch,How should I do? thanks! andy 2009-04-02 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/5a03ddc8/attachment.html From yivzhenko at mksat.net Thu Apr 2 03:22:05 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko (WP)) Date: Thu, 2 Apr 2009 13:22:05 +0300 Subject: [Freeswitch-users] Use rates from lcr in nibblebill module Message-ID: <200904021322.05690.yivzhenko@mksat.net> Hi, I want to use module lcr to find a best route and his rate , then make a call and bill on that rate with nibblebill module. lcr return variable "lcr_auto_route" that contains "[lcr_rate=xxx]" variable for new channel. To use nibblebill i need to set "nibble_rate" = "lcr_rate". What is best method to do that? Is there a way to do that with standard tools, without use external scripts? Thanks, Yuriy From brian at freeswitch.org Thu Apr 2 04:58:58 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 06:58:58 -0500 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <49D481CC.70102@gcdf.pl> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> Message-ID: <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> On Apr 2, 2009, at 4:13 AM, Szymon Olko wrote: > I did not think yet about HA nor LB. > > I tested how FS handles high load. All my calls are placed in > mod_conference. When cpu usage gets it's limits then new calls can > be placed but sound quality is getting worst with every next call. > When calls are hanged up then sound gets better. I did not test > it to see what happens when more and more calls are created. > FS has very low memory consumption and I think that CPU is the > limit. I did not notice any monitoring of CPU usage by FS, but my > installation is limited to only few modules, so maybe I'm missing > something. Load testing against the conference module is about the worst thing you can do... tossing 100+ people in the same conference isn't going to scale well for load testing because its not something you usually do in a real world scenario. Usually you'll have most of the participants muted. I highly recommend you try doing something like a bridge or a file playback from a ram disk. >> >> Would love to hear some experiences of deploying FS with failover & >> high > My failover is currently made by shell script which every 10 seconds > check for working FS and restarts it if it does not work. > I use svn trunk so crash happens once a while, but they are > successfully fixed by developers. > > Once there was a problem that conference module was stuck and did > not respond to my commands. I made script with netcat which > checks once a while for response and restarts if there is none. >> load. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/44622516/attachment.html From rupa at rupa.com Thu Apr 2 05:37:28 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 2 Apr 2009 07:37:28 -0500 Subject: [Freeswitch-users] Use rates from lcr in nibblebill module In-Reply-To: <200904021322.05690.yivzhenko@mksat.net> References: <200904021322.05690.yivzhenko@mksat.net> Message-ID: Update the to the latest. I've added more channel vars: eg: after doing: (not a real number) I get the following: variable_lcr_query_digits: [12148267722] variable_lcr_query_profile: [0] variable_lcr_query_expanded_digits: [12148267722, 1214826772, 121482677, 12148267, 1214826, 121482, 12148, 1214, 121, 12, 1] variable_lcr_route_1: [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gateway/teliax/12148267722] variable_lcr_rate_1: [0.01000] variable_lcr_carrier_1: [teliax] variable_lcr_codec_1: [PCMU] variable_lcr_route_2: [[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec_string=PCMU]sofia/gateway/vitelity/12148267722] variable_lcr_rate_2: [0.01440] variable_lcr_carrier_2: [vitelity] variable_lcr_codec_2: [PCMU] variable_lcr_route_count: [2] variable_lcr_auto_route: [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gateway/teliax/12148267722|[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec_string=PCMU]sofia/gateway/vitelity/12148267722] variable_import: [lcr_carrier,lcr_rate] which, I think is what you are asking for. If you know which route you are going to use (eg: 1) then you can get it's rate by using lcr_rate_1. Alternatively, you can use the lcr_auto_route and then once the b-leg connects, query the b-leg variable for lcr_carrier and lcr_rate to see which one was actually used. You really can't use lcr_auto_route and set a single rate since each leg can be rated differently (look at example above). Normally lcr is used for your own rates between you and your carrier. That is independant of the rate table used for your customers. You can use lcr to query both. First use lcr to query your own rates using a different profile. This would return a *single* route if you setup your route table right. Save the rate in a var to be used with nibble bill. Then use lcr with your external rates so you can route according to your own cost with your carrier(s). On Thu, Apr 2, 2009 at 5:22 AM, Yuriy Ivzhenko (WP) wrote: > Hi, > > I want to use module lcr to find a best route and his rate , then make a > call > and bill on that rate with nibblebill module. > > lcr return variable "lcr_auto_route" that contains "[lcr_rate=xxx]" > variable > for new channel. > To use nibblebill i need to set "nibble_rate" = "lcr_rate". > > What is best method to do that? > Is there a way to do that with standard tools, without use external > scripts? > > > Thanks, > Yuriy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090402/bc23b3e1/attachment.html From bipin at xbipin.com Thu Apr 2 06:01:57 2009 From: bipin at xbipin.com (xbipin) Date: Thu, 2 Apr 2009 06:01:57 -0700 (PDT) Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> Message-ID: <22847331.post@talk.nabble.com> hi, i wanted to know if there was any way to actually accept all registrations coming towards freeswitch, the normal function is to have all the suerid and passwords configured, but is there a way to accept all registrations coming towards a single ip or domain? Regards, Bipin -- View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Prometheus001 at gmx.net Thu Apr 2 06:08:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Apr 2009 15:08:25 +0200 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: <7C5053E3-1F30-4D6C-B786-004931E813A1@freeswitch.org> References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> <35b355e90904011759s43c91d4cg4d7a95b66d8027bd@mail.gmail.com> <7C5053E3-1F30-4D6C-B786-004931E813A1@freeswitch.org> Message-ID: <49D4B8C9.6070401@gmx.net> Wow, this is cool. Fantastic work! I tried this immediately. This is also very useful to share data across applications. Here an example how to share data between Freeswitch and a ruby memcache-client: On Ruby/Rails I set the namespace e.g. to "freeswitch" for the same memcached server in environment.rb In Freeswitch I add the following line to the dialplan: Take care to prefix your key (here "test") with the Ruby namespace "freeswitch:" Now you can receive the data in Ruby in raw mode: >> CACHE.get("test",0) => 'This is a test" The 0 as second parameter is important for the raw mode, otherwise ruby will try to marshall the result from memcached and fails. I added this info to the wiki. Best regards Peter Brian West schrieb: > At the very least you didn't say "I can't wait to play with it!" :P > > > On Apr 1, 2009, at 7:59 PM, Shelby Ramsey wrote: > >> Rupa, >> >> This is a big contribution! Thanks! Can't wait to play with this. >> >> SDR > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 leon at scarlet-internet.nl Thu Apr 2 06:13:01 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 2 Apr 2009 15:13:01 +0200 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <22847331.post@talk.nabble.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> Message-ID: <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> Hi, You can blindly accept registrations and / or authentication messages with these parameters in a sip profile: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#accept-blind-reg regards, Leon On Apr 2, 2009, at 3:01 PM, xbipin wrote: > > hi, > > i wanted to know if there was any way to actually accept all > registrations > coming towards freeswitch, the normal function is to have all the > suerid and > passwords configured, but is there a way to accept all registrations > coming > towards a single ip or domain? > > > Regards, > Bipin > -- > View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.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 rupa at rupa.com Thu Apr 2 06:13:23 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 2 Apr 2009 08:13:23 -0500 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: <49D4B8C9.6070401@gmx.net> References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> <35b355e90904011759s43c91d4cg4d7a95b66d8027bd@mail.gmail.com> <7C5053E3-1F30-4D6C-B786-004931E813A1@freeswitch.org> <49D4B8C9.6070401@gmx.net> Message-ID: On Thu, Apr 2, 2009 at 8:08 AM, Peter P GMX wrote: > Wow, this is cool. Fantastic work! > I tried this immediately. This is also very useful to share data across > applications. > [snip] > > I added this info to the wiki. > > Best regards > Peter > Thanks for the wiki update -- great to see examples of how to actually use it. :) -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/28e96090/attachment.html From anthony.minessale at gmail.com Thu Apr 2 06:51:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Apr 2009 08:51:16 -0500 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: References: Message-ID: <191c3a030904020651ideeb958v77be91490a6ff38f@mail.gmail.com> Its the buffering and startup of the shout stream taking up the time, HINT put the shoutcast stream into a local stream with a .loc file and then play that in the conference. 2009/4/1 Rupa Schomaker > I've setup a conference bridge that has perpetual-sound set to a icecast > stream. When the first person connects, there is an ~7s delay before any > audio is heard. This is similar to a problem reported by Dan here and > concluded with Tony adding the channel var "enable_file_write_buffering". > The list discussion ended here: > http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011104.html > > > I set this var in my dialplan: > > > prior to joining the conference. > > The first person in still sees a 7s delay on audio the first time in. > > Like dan, I have icecast setup with > burst_on_connect set to 1 > but my burst_size is the default 64k so quite a bit of data. > > Has anyone been able to get an on-demand shoutcast stream from an icecast > server to start immediately (or at least within a second)? > > Thanks. > > -- > -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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/4a888955/attachment.html From rupa at rupa.com Thu Apr 2 07:05:38 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 2 Apr 2009 09:05:38 -0500 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <191c3a030904020651ideeb958v77be91490a6ff38f@mail.gmail.com> References: <191c3a030904020651ideeb958v77be91490a6ff38f@mail.gmail.com> Message-ID: 2009/4/2 Anthony Minessale > Its the buffering and startup of the shout stream taking up the time, > > HINT put the shoutcast stream into a local stream with a .loc file and then > play that in the conference. > Ah, that is easy enough! Though I think with icecast doing the burst_on_connect thingie there should be enough data (pushed much faster than real time) to fill FS's buffers. But that would require mod_shout to cooperate with that strategy. ie: on connect, drain the socket as fast as it can filling it's own buffers. Once it's own buffers are full start streaming. I think right now it drains the socket only as fast as it needs to. Or maybe not. I'll go the local stream route for now.... -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/0372a030/attachment.html From Prometheus001 at gmx.net Thu Apr 2 07:05:51 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Apr 2009 16:05:51 +0200 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> Message-ID: <49D4C63F.8050400@gmx.net> I use the access control list acl.conf.xml to configure that. Put ip/mask into the domain part of this config file, then it accepts calls from this ip (eg. ip/32.) or from this subnet (e.g. ip/24). Best regards Peter Leon de Rooij schrieb: > Hi, > > You can blindly accept registrations and / or authentication messages > with these parameters in a sip profile: > > > > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#accept-blind-reg > > regards, > > Leon > > On Apr 2, 2009, at 3:01 PM, xbipin wrote: > > >> hi, >> >> i wanted to know if there was any way to actually accept all >> registrations >> coming towards freeswitch, the normal function is to have all the >> suerid and >> passwords configured, but is there a way to accept all registrations >> coming >> towards a single ip or domain? >> >> >> Regards, >> Bipin >> -- >> View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.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 >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mike at jerris.com Thu Apr 2 07:07:53 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 2 Apr 2009 10:07:53 -0400 Subject: [Freeswitch-users] about freeswitch conference In-Reply-To: <200904021529137966712@gmail.com> References: <200904021524116567464@gmail.com> <200904021529137966712@gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Mod_conference On Apr 2, 2009, at 3:29 AM, bmsword wrote: > I want to use another softswitch conference that has been > deployed in freeswitch,How should I do? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/804ee06f/attachment.html From bipin at xbipin.com Thu Apr 2 07:38:35 2009 From: bipin at xbipin.com (Bipin Patel) Date: Thu, 02 Apr 2009 18:38:35 +0400 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <49D4C63F.8050400@gmx.net> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> Message-ID: <49D4CDEB.1040201@xbipin.com> hi, will the below work if all the registration that is to be accepted come from different public ip addresses, i mean, clients from all ip ranges and addresses rather than a single ip Regards, Bipin www.xbipin.com +971-55-9270058 -------- Original Message -------- Subject: Re: [Freeswitch-users] upper registration in FS? From: Peter P GMX To: freeswitch-users at lists.freeswitch.org Date: Thursday, April 02, 2009 6:05:51 PM > I use the access control list acl.conf.xml to configure that. > > Put ip/mask into the domain part of this config file, then it accepts > calls from this ip (eg. ip/32.) or from this subnet (e.g. ip/24). > > > > > > > > Best regards > Peter > > Leon de Rooij schrieb: >> Hi, >> >> You can blindly accept registrations and / or authentication messages >> with these parameters in a sip profile: >> >> >> >> >> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#accept-blind-reg >> >> regards, >> >> Leon >> >> On Apr 2, 2009, at 3:01 PM, xbipin wrote: >> >> >>> hi, >>> >>> i wanted to know if there was any way to actually accept all >>> registrations >>> coming towards freeswitch, the normal function is to have all the >>> suerid and >>> passwords configured, but is there a way to accept all registrations >>> coming >>> towards a single ip or domain? >>> >>> >>> Regards, >>> Bipin >>> -- >>> View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.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 >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > __________ NOD32 3983 (20090402) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > > > From intralanman at freeswitch.org Thu Apr 2 07:50:21 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 02 Apr 2009 10:50:21 -0400 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <49D4CDEB.1040201@xbipin.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> <49D4CDEB.1040201@xbipin.com> Message-ID: <49D4D0AD.9030904@freeswitch.org> Bipin Patel wrote: > hi, > > will the below work if all the registration that is to be accepted come > from different public ip addresses, i mean, clients from all ip ranges > and addresses rather than a single ip > yeah, that's kinda why its called "blind" ... you don't have to know where its coming from, and it doesn't have to be valid... just "blindly" accepts it -Ray From Richard.Lamkin at mettoni.com Thu Apr 2 08:12:22 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 2 Apr 2009 16:12:22 +0100 Subject: [Freeswitch-users] Database schema Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804BE74B1@nickel.mettonigroup.com> Are there documents or wiki page [I've missed during my searches] that detail the records and their types that are stored in the various FS databases; e.g. sofia_reg_.db, core.db ? Regards Richard Lamkin ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/f99c8aa5/attachment.html From brian at freeswitch.org Thu Apr 2 08:21:44 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 10:21:44 -0500 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <49D4CDEB.1040201@xbipin.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> <49D4CDEB.1040201@xbipin.com> Message-ID: <27317F98-DDB5-49D3-93B8-F0B8949710FB@freeswitch.org> Turn on Multireg too. /b On Apr 2, 2009, at 9:38 AM, Bipin Patel wrote: > hi, > > will the below work if all the registration that is to be accepted > come > from different public ip addresses, i mean, clients from all ip ranges > and addresses rather than a single ip > > > > > Regards, > Bipin > www.xbipin.com > +971-55-9270058 Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/9245e267/attachment.html From mike at jerris.com Thu Apr 2 08:28:32 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 2 Apr 2009 11:28:32 -0400 Subject: [Freeswitch-users] Database schema In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804BE74B1@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804BE74B1@nickel.mettonigroup.com> Message-ID: <6C26E3C4-1230-4BE7-A6DD-A5B4ECADBD95@jerris.com> no, but they all auto-create. You can create a db and set up odbc, start freeswitch, then dump your db schema. Also, please do not send confidential emails to the mailing list. Mike On Apr 2, 2009, at 11:12 AM, Richard Lamkin wrote: > Are there documents or wiki page [I?ve missed during my searches] > that detail the records and their types that are stored in the > various FS databases; e.g. sofia_reg_.db, core.db ? > > Regards > Richard Lamkin > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/b5cb06b2/attachment.html From bipin at xbipin.com Thu Apr 2 08:40:14 2009 From: bipin at xbipin.com (Bipin Patel) Date: Thu, 02 Apr 2009 19:40:14 +0400 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <27317F98-DDB5-49D3-93B8-F0B8949710FB@freeswitch.org> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> <49D4CDEB.1040201@xbipin.com> <27317F98-DDB5-49D3-93B8-F0B8949710FB@freeswitch.org> Message-ID: <49D4DC5E.4080506@xbipin.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/ff02f04e/attachment.html From cstomi.levlist at gmail.com Thu Apr 2 09:46:57 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Thu, 02 Apr 2009 18:46:57 +0200 Subject: [Freeswitch-users] loopback-b channels stay alive Message-ID: <49D4EC01.6050205@gmail.com> Hello, We originate loopback channels and they end up in calling sofia and transfer the call to a fifo. If we have a heavy call volume loopback-b channels don't hangup properly. They stay in core.db. Unfortunetly we can't reproduce it on test boxes but happens every day. On this box we had to turn off debug logging, becase we had I/O problems. The only thing I saw in log that switch_core_session_thread don't call switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n", session->id, switch_channel_get_name(session->channel)); in these cases. We have local patches (I don't think they are related) and we are running FS on virtual machine and we had some problem with that before so I'm not sure, but I guess it is maybe a lock or mutex problem. I tried SWITCH_DEBUG_RWLOCKS, but I got build error, and I don't know what to do with it. FS_CFLAGS = -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS export CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" export MOD_CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" ./configure gcc -I/DEVEL/freeswitch/src/include -I/DEVEL/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS -Wall -std=c99 -pedantic -o .libs/freeswitch freeswitch-switch.o -lm ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath -Wl,/opt/freeswitch//lib ./.libs/libfreeswitch.so: undefined reference to `switch_core_session_read_lock' ./.libs/libfreeswitch.so: undefined reference to `switch_core_session_locate' ./.libs/libfreeswitch.so: undefined reference to `switch_core_session_rwunlock' collect2: ld returned 1 exit status make[2]: *** [freeswitch] Error 1 Could you please tell me how could I test mutexes, rwlocks? Other option would be to omit loopback channels. Anthony earlier suggested me to avoid it and call sofia directly "you could make the loopback channel execute the eval app and do the originate to the sofia channel from the dialplan. or make the loopback chan exec a lua or js and fire an originate command and exit This way you don't have the loopback a and b leg as well as the sofia chan." but it doesn't work, because originate api doesn't let us originate inside a session. So we still using it. Thanks in advance, Tamas From msc at freeswitch.org Thu Apr 2 10:07:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 2 Apr 2009 10:07:18 -0700 Subject: [Freeswitch-users] loopback-b channels stay alive In-Reply-To: <49D4EC01.6050205@gmail.com> References: <49D4EC01.6050205@gmail.com> Message-ID: <87f2f3b90904021007j1d2ae759n388d05078c826219@mail.gmail.com> Thanks for doing some of the legwork on this. BTW, this thread is probably a bit too technical for the users list - I recommend sending to the dev list. :) -MC On Thu, Apr 2, 2009 at 9:46 AM, Tamas Cseke wrote: > Hello, > > We originate loopback channels and they end up in calling sofia > and transfer the call to a fifo. > > If we have a heavy call volume loopback-b channels don't hangup properly. > They stay in core.db. > Unfortunetly we can't reproduce it on test boxes but happens every day. > On this box we had to turn off debug logging, becase we had I/O problems. > > The only thing I saw in log that switch_core_session_thread don't call > > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Session %" > SWITCH_SIZE_T_FMT " (%s) Ended\n", > session->id, > switch_channel_get_name(session->channel)); > > in these cases. > We have local patches (I don't think they are related) and we are > running FS on virtual machine and we had some problem with that before > so I'm not sure, but I guess it is maybe a lock or mutex problem. > > I tried SWITCH_DEBUG_RWLOCKS, but I got build error, and I don't know > what to do with it. > > FS_CFLAGS = -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS > export CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" > export MOD_CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" > ./configure > > gcc -I/DEVEL/freeswitch/src/include > -I/DEVEL/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS -Wall -std=c99 > -pedantic -o .libs/freeswitch freeswitch-switch.o -lm > ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt > -lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath > -Wl,/opt/freeswitch//lib > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_read_lock' > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_locate' > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_rwunlock' > collect2: ld returned 1 exit status > make[2]: *** [freeswitch] Error 1 > > Could you please tell me how could I test mutexes, rwlocks? > > Other option would be to omit loopback channels. > Anthony earlier suggested me to avoid it and call sofia directly > > "you could make the loopback channel execute the eval app and do the > originate to the sofia channel from the dialplan. > > > or make the loopback chan exec a lua or js and fire an originate command > and > exit > > This way you don't have the loopback a and b leg as well as the sofia > chan." > > but it doesn't work, because originate api doesn't let us originate inside > a session. > So we still using it. > > > Thanks in advance, > Tamas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/3e1381c5/attachment.html From brian at freeswitch.org Thu Apr 2 10:20:31 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 12:20:31 -0500 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <2d9149cd0904012309h5428ece5qce9a287b0972dd81@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <8b3b7acc0904011641v2c7e3768ne72e06623ed064f2@mail.gmail.com> <2d9149cd0904012309h5428ece5qce9a287b0972dd81@mail.gmail.com> Message-ID: hehe I emailed it to him off list :) /b On Apr 2, 2009, at 1:09 AM, Kristian Kielhofner wrote: > I probably shouldn't be doing this for you, but... > > http://bugs.digium.com/view.php?id=14431 > > ;) Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/c8a23c63/attachment.html From anthony.minessale at gmail.com Thu Apr 2 11:03:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Apr 2009 13:03:51 -0500 Subject: [Freeswitch-users] loopback-b channels stay alive In-Reply-To: <49D4EC01.6050205@gmail.com> References: <49D4EC01.6050205@gmail.com> Message-ID: <191c3a030904021103o40307040xfd5489763644ea72@mail.gmail.com> you can't pass it in with -D you have to actually add #define SWITCH_DEBUG_RWLOCKS to the top of switch_core.h On Thu, Apr 2, 2009 at 11:46 AM, Tamas Cseke wrote: > Hello, > > We originate loopback channels and they end up in calling sofia > and transfer the call to a fifo. > > If we have a heavy call volume loopback-b channels don't hangup properly. > They stay in core.db. > Unfortunetly we can't reproduce it on test boxes but happens every day. > On this box we had to turn off debug logging, becase we had I/O problems. > > The only thing I saw in log that switch_core_session_thread don't call > > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Session %" > SWITCH_SIZE_T_FMT " (%s) Ended\n", > session->id, > switch_channel_get_name(session->channel)); > > in these cases. > We have local patches (I don't think they are related) and we are > running FS on virtual machine and we had some problem with that before > so I'm not sure, but I guess it is maybe a lock or mutex problem. > > I tried SWITCH_DEBUG_RWLOCKS, but I got build error, and I don't know > what to do with it. > > FS_CFLAGS = -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS > export CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" > export MOD_CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" > ./configure > > gcc -I/DEVEL/freeswitch/src/include > -I/DEVEL/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS -Wall -std=c99 > -pedantic -o .libs/freeswitch freeswitch-switch.o -lm > ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt > -lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath > -Wl,/opt/freeswitch//lib > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_read_lock' > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_locate' > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_rwunlock' > collect2: ld returned 1 exit status > make[2]: *** [freeswitch] Error 1 > > Could you please tell me how could I test mutexes, rwlocks? > > Other option would be to omit loopback channels. > Anthony earlier suggested me to avoid it and call sofia directly > > "you could make the loopback channel execute the eval app and do the > originate to the sofia channel from the dialplan. > > > or make the loopback chan exec a lua or js and fire an originate command > and > exit > > This way you don't have the loopback a and b leg as well as the sofia > chan." > > but it doesn't work, because originate api doesn't let us originate inside > a session. > So we still using it. > > > Thanks in advance, > Tamas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/3090d55b/attachment.html From solko at gcdf.pl Thu Apr 2 12:29:28 2009 From: solko at gcdf.pl (Szymon Olko) Date: Thu, 02 Apr 2009 21:29:28 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> Message-ID: <49D51218.2080209@gcdf.pl> Brian West pisze: > > On Apr 2, 2009, at 4:13 AM, Szymon Olko wrote: > >> I did not think yet about HA nor LB. >> >> I tested how FS handles high load. All my calls are placed in >> mod_conference. When cpu usage gets it's limits then new calls can >> be placed but sound quality is getting worst with every next call. >> When calls are hanged up then sound gets better. I did not test >> it to see what happens when more and more calls are created. >> FS has very low memory consumption and I think that CPU is the limit. >> I did not notice any monitoring of CPU usage by FS, but my >> installation is limited to only few modules, so maybe I'm missing >> something. > > Load testing against the conference module is about the worst thing you > can do... tossing 100+ people in the same conference isn't going to > scale well for load testing because its not something you usually do in > a real world scenario. Usually you'll have most of the participants muted. > > I highly recommend you try doing something like a bridge or a file > playback from a ram disk. > I did not described it perfectly. I made agents, queues scenarios on conferences. This what I tested was for example 100 calls, so it's 200 channels, and 100 conferences, 2 channels per conference, all are unmuted. I did that just because it is my work scenario. >>> >>> Would love to hear some experiences of deploying FS with failover & high >> My failover is currently made by shell script which every 10 seconds >> check for working FS and restarts it if it does not work. >> I use svn trunk so crash happens once a while, but they are >> successfully fixed by developers. >> >> Once there was a problem that conference module was stuck and did not >> respond to my commands. I made script with netcat which >> checks once a while for response and restarts if there is none. >>> load. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 2 12:34:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 14:34:51 -0500 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <49D51218.2080209@gcdf.pl> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> <49D51218.2080209@gcdf.pl> Message-ID: <2CD5CE30-E2E5-4D72-A587-19F92D862665@freeswitch.org> what kind of hardware? /b On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: > I did not described it perfectly. I made agents, queues scenarios on > conferences. > This what I tested was for example 100 calls, so it's 200 channels, > and 100 conferences, 2 channels per conference, all are > unmuted. I did that just because it is my work scenario. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/c6244405/attachment-0001.html From Prometheus001 at gmx.net Thu Apr 2 13:07:10 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Apr 2009 22:07:10 +0200 Subject: [Freeswitch-users] Handle invite with wrong to:IP Message-ID: <49D51AEE.7010904@gmx.net> Hello, I am using a SIP account from Netvoip CH. I try to receive inbound call from this SIP trunk. I discovered that, when they sent an invite, the IP-Adress of the to: is their own IP address. There fore ACL doesn't work and FS asks for authorization, which then fails I receive the following message on the CLI: 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() Can't find user [anonymous at 62.65.128.62] You must define a domain called '62.65.128.62' in your directory and add a user with the id="anonymous" attribute and you must configure your device to use the proper domain in it's authentication credentials. I could do that, but this is not clean and I do not have a password for that. How can I workaround this, so that Freeswitch accepts this call? Aliases do not seem to work. Here is a sample message after FS asks for authorization: xx.xx.xxx.xxx is the IP of our Freeswitch 62.65.128.62 is the IP of Netvoip CH I would expect To: . instead of To: . U 62.65.128.62:5060 -> xx.xx.xxx.xxx:5080 INVITE sip:0715aaaaaa at xx.xx.xxx.xxx:5080 SIP/2.0. Via: SIP/2.0/UDP 62.65.128.62:5060. Via: SIP/2.0/UDP 62.65.128.61:5060;branch=z9hG4bK8c977d2613c4d7d1fd9d03d4. Max-Forwards: 69. From: ;tag=8c977d2613672832fd9d03e9. To: . Call-ID: 8c977d261329cd80fd9d03d6 at 62.65.128.61. CSeq: 2 INVITE. User-agent: Netstream VoIP Gateway. Contact: . Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 584. Proxy-Authorization: Digest username="anonymous", realm="62.65.128.62", nonce="a4151ee0-1fbb-11de-b056-494b9de21e06", nc="00000001", uri="sip:0715aaaaaa at 62.65.128.62:5060", cnonce="5f109eee", response="62faa6d38b3b12c3626753395a8b507c", algorithm="MD5", qop="auth". . v=0. o=- 225947743692042 1 IN IP4 62.65.128.62. s=-. c=IN IP4 62.65.128.62. t=0 0. m=audio 28224 RTP/AVP 8 18 4 3 100 100 99 100 100 98 97 96 105 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=no. a=rtpmap:3 GSM/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:99 G726-16/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:97 G726-32/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:105 iLBC/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. Best regards Peter From solko at gcdf.pl Thu Apr 2 13:07:53 2009 From: solko at gcdf.pl (Szymon Olko) Date: Thu, 02 Apr 2009 22:07:53 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <2CD5CE30-E2E5-4D72-A587-19F92D862665@freeswitch.org> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> <49D51218.2080209@gcdf.pl> <2CD5CE30-E2E5-4D72-A587-19F92D862665@freeswitch.org> Message-ID: <49D51B19.3050709@gcdf.pl> Brian West pisze: > what kind of hardware? > I made testes on Pentium-M laptop with single core 1,6Hz. I did not write those results, it was over 100 calls that was handle good, I was just curios what will happen. Tomorrow I will make real testes. My production works on 2 core P4 and I have there only 35 agents CPU load is like 7% with 15% small peeks. All phones are sip or analog via sip gateways, PRI is currently still on asterisk which is connected via sip. > /b > > On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: > >> I did not described it perfectly. I made agents, queues scenarios on >> conferences. >> This what I tested was for example 100 calls, so it's 200 channels, >> and 100 conferences, 2 channels per conference, all are >> unmuted. I did that just because it is my work scenario. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 ceino.no at gmail.com Thu Apr 2 12:55:25 2009 From: ceino.no at gmail.com (Ceino) Date: Thu, 02 Apr 2009 21:55:25 +0200 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre3 Now Available In-Reply-To: <87f2f3b90904010949x4d3fc2f8ia9feda2581956f68@mail.gmail.com> References: <87f2f3b90904010949x4d3fc2f8ia9feda2581956f68@mail.gmail.com> Message-ID: <49D5182D.1080508@gmail.com> Hi, I have tested it a little bit and it's works well. But when I give it the command to stop (...) it use about 40 sec. to stop (1.0.3 use about 5 sec). Here is a log over where is hang (looks like a Sofia thread use long time to stop): ------------------------------------------------------------------ 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal-ipv6 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 sofia_glue_del_profile() deleted gateway example.com 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock external ------------------------------------------------------------------ Best Regards Lars Sivertsen Michael Collins wrote: > The FreeSWITCH team would like to let everyone know that the latest > version is available. More information can be found here: > http://www.freeswitch.org/node/172 > > By all means download, upgrade, test, and report back! Your feedback > helps us make FreeSWITCH even better! > -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 > From ceino.no at gmail.com Thu Apr 2 13:04:28 2009 From: ceino.no at gmail.com (Ceino) Date: Thu, 02 Apr 2009 22:04:28 +0200 Subject: [Freeswitch-users] FreeSWITCH 1.0.4pre3 is slow to stop Message-ID: <49D51A4C.7040701@gmail.com> Hi all, I have tested it a little bit and it's works well. But when I give it the command to stop (...) it use about 40 sec to stop (1.0.3 use about 5 sec). Here is a log over where is hang (looks like a Sofia thread use long time to stop): ------------------------------------------------------------------ 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal-ipv6 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 sofia_glue_del_profile() deleted gateway example.com 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock external ------------------------------------------------------------------ Best Regards Lars Sivertsen From brian at freeswitch.org Thu Apr 2 13:11:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 15:11:22 -0500 Subject: [Freeswitch-users] FreeSWITCH 1.0.4pre3 is slow to stop In-Reply-To: <49D51A4C.7040701@gmail.com> References: <49D51A4C.7040701@gmail.com> Message-ID: <73F10AE0-6EDA-4F02-A4D9-BA8AF73CB070@freeswitch.org> Try updating to SVN trunk... I think we fixed that already. /b On Apr 2, 2009, at 3:04 PM, Ceino wrote: > Hi all, I have tested it a little bit and it's works well. But when I > give it the command to stop (...) > it use about 40 sec to stop (1.0.3 use about 5 sec). > > Here is a log over where is hang (looks like a Sofia thread use long > time to stop): > ------------------------------------------------------------------ > > 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() > Waiting for worker thread > 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() > Write > unlock internal-ipv6 > 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() > Write > unlock internal > 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() > Waiting for worker thread > 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 > sofia_glue_del_profile() > deleted gateway example.com > 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() > Write > unlock external > ------------------------------------------------------------------ > > > Best Regards > > Lars Sivertsen Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/4283c560/attachment.html From brian at freeswitch.org Thu Apr 2 13:14:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 15:14:42 -0500 Subject: [Freeswitch-users] Handle invite with wrong to:IP In-Reply-To: <49D51AEE.7010904@gmx.net> References: <49D51AEE.7010904@gmx.net> Message-ID: We use the true network ip the invite came from NOT the one in the sip headers. Not very trust worth to do that you think? ;) So if your ACL is correctly setup to 62.65.128.62 it would let them in please verify your ACL is correct... /b On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote: > Hello, > > I am using a SIP account from Netvoip CH. I try to receive inbound > call > from this SIP trunk. I discovered that, when they sent an invite, the > IP-Adress of the to: is their own IP address. > There fore ACL doesn't work and FS asks for authorization, which > then fails > > I receive the following message on the CLI: > 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() > Can't find user [anonymous at 62.65.128.62] > You must define a domain called '62.65.128.62' in your directory and > add > a user with the id="anonymous" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/5fa2980e/attachment.html From anthony.minessale at gmail.com Thu Apr 2 13:14:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Apr 2009 15:14:53 -0500 Subject: [Freeswitch-users] Handle invite with wrong to:IP In-Reply-To: <49D51AEE.7010904@gmx.net> References: <49D51AEE.7010904@gmx.net> Message-ID: <191c3a030904021314o461ef854hcf856be9f406f38e@mail.gmail.com> acl uses the remote addr from the socket connection, not anything from the sip message. On Thu, Apr 2, 2009 at 3:07 PM, Peter P GMX wrote: > Hello, > > I am using a SIP account from Netvoip CH. I try to receive inbound call > from this SIP trunk. I discovered that, when they sent an invite, the > IP-Adress of the to: is their own IP address. > There fore ACL doesn't work and FS asks for authorization, which then fails > > I receive the following message on the CLI: > 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() > Can't find user [anonymous at 62.65.128.62] > You must define a domain called '62.65.128.62' in your directory and add > a user with the id="anonymous" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > > I could do that, but this is not clean and I do not have a password for > that. > > How can I workaround this, so that Freeswitch accepts this call? Aliases > do not seem to work. > > Here is a sample message after FS asks for authorization: > xx.xx.xxx.xxx is the IP of our Freeswitch > 62.65.128.62 is the IP of Netvoip CH > > I would expect > To: . > instead of > To: >. > > U 62.65.128.62:5060 -> xx.xx.xxx.xxx:5080 > INVITE sip:0715aaaaaa at xx.xx.xxx.xxx:5080 SIP/2.0. > Via: SIP/2.0/UDP 62.65.128.62:5060. > Via: SIP/2.0/UDP 62.65.128.61:5060;branch=z9hG4bK8c977d2613c4d7d1fd9d03d4. > Max-Forwards: 69. > From: > >;tag=8c977d2613672832fd9d03e9. > To: >. > Call-ID: 8c977d261329cd80fd9d03d6 at 62.65.128.61. > CSeq: 2 INVITE. > User-agent: Netstream VoIP Gateway. > Contact: . > Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE. > Content-Type: application/sdp. > Content-Length: 584. > Proxy-Authorization: Digest username="anonymous", realm="62.65.128.62", > nonce="a4151ee0-1fbb-11de-b056-494b9de21e06", nc="00000001", > uri="sip:0715aaaaaa at 62.65.128.62:5060", cnonce="5f109eee", > response="62faa6d38b3b12c3626753395a8b507c", algorithm="MD5", qop="auth". > . > v=0. > o=- 225947743692042 1 IN IP4 62.65.128.62. > s=-. > c=IN IP4 62.65.128.62. > t=0 0. > m=audio 28224 RTP/AVP 8 18 4 3 100 100 99 100 100 98 97 96 105 0 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:4 G723/8000. > a=fmtp:4 annexa=no. > a=rtpmap:3 GSM/8000. > a=rtpmap:100 speex/8000. > a=rtpmap:100 speex/8000. > a=rtpmap:99 G726-16/8000. > a=rtpmap:100 speex/8000. > a=rtpmap:100 speex/8000. > a=rtpmap:98 G726-24/8000. > a=rtpmap:97 G726-32/8000. > a=rtpmap:96 G726-40/8000. > a=rtpmap:105 iLBC/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/fbea3fad/attachment-0001.html From anthony.minessale at gmail.com Thu Apr 2 13:15:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Apr 2009 15:15:58 -0500 Subject: [Freeswitch-users] FreeSWITCH 1.0.4pre3 is slow to stop In-Reply-To: <49D51A4C.7040701@gmail.com> References: <49D51A4C.7040701@gmail.com> Message-ID: <191c3a030904021315m130ab671t9e0c94f4bf7973e9@mail.gmail.com> wait for pre4 On Thu, Apr 2, 2009 at 3:04 PM, Ceino wrote: > Hi all, I have tested it a little bit and it's works well. But when I > give it the command to stop (...) > it use about 40 sec to stop (1.0.3 use about 5 sec). > > Here is a log over where is hang (looks like a Sofia thread use long > time to stop): > ------------------------------------------------------------------ > > 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() > Waiting for worker thread > 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write > unlock internal-ipv6 > 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write > unlock internal > 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() > Waiting for worker thread > 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 sofia_glue_del_profile() > deleted gateway example.com > 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write > unlock external > ------------------------------------------------------------------ > > > Best Regards > > Lars Sivertsen > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/89631066/attachment.html From Prometheus001 at gmx.net Thu Apr 2 13:34:10 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Apr 2009 22:34:10 +0200 Subject: [Freeswitch-users] Handle invite with wrong to:IP In-Reply-To: References: <49D51AEE.7010904@gmx.net> Message-ID: <49D52142.7040401@gmx.net> My ACL contains: So this should be fine, right? However it doesn't work. Best regards Peter Brian West schrieb: > We use the true network ip the invite came from NOT the one in the sip > headers. Not very trust worth to do that you think? ;) > > So if your ACL is correctly setup to 62.65.128.62 it would let them in > please verify your ACL is correct... > > /b > > On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote: > >> Hello, >> >> I am using a SIP account from Netvoip CH. I try to receive inbound call >> from this SIP trunk. I discovered that, when they sent an invite, the >> IP-Adress of the to: is their own IP address. >> There fore ACL doesn't work and FS asks for authorization, which then >> fails >> >> I receive the following message on the CLI: >> 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() >> Can't find user [anonymous at 62.65.128.62 ] >> You must define a domain called '62.65.128.62' in your directory and add >> a user with the id="anonymous" attribute >> and you must configure your device to use the proper domain in it's >> authentication credentials. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 2 14:24:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Apr 2009 16:24:29 -0500 Subject: [Freeswitch-users] Handle invite with wrong to:IP In-Reply-To: <49D52142.7040401@gmx.net> References: <49D51AEE.7010904@gmx.net> <49D52142.7040401@gmx.net> Message-ID: <191c3a030904021424ta0f4adwfe805f9d9bda86e@mail.gmail.com> look at the debug log and see what happens? On Thu, Apr 2, 2009 at 3:34 PM, Peter P GMX wrote: > My ACL contains: > > > > > > So this should be fine, right? However it doesn't work. > > Best regards > Peter > > > Brian West schrieb: > > We use the true network ip the invite came from NOT the one in the sip > > headers. Not very trust worth to do that you think? ;) > > > > So if your ACL is correctly setup to 62.65.128.62 it would let them in > > please verify your ACL is correct... > > > > /b > > > > On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote: > > > >> Hello, > >> > >> I am using a SIP account from Netvoip CH. I try to receive inbound call > >> from this SIP trunk. I discovered that, when they sent an invite, the > >> IP-Adress of the to: is their own IP address. > >> There fore ACL doesn't work and FS asks for authorization, which then > >> fails > >> > >> I receive the following message on the CLI: > >> 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() > >> Can't find user [anonymous at 62.65.128.62 >] > >> You must define a domain called '62.65.128.62' in your directory and add > >> a user with the id="anonymous" attribute > >> and you must configure your device to use the proper domain in it's > >> authentication credentials. > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/c2902cb0/attachment.html From Prometheus001 at gmx.net Thu Apr 2 14:45:40 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Apr 2009 23:45:40 +0200 Subject: [Freeswitch-users] Handle invite with wrong to:IP In-Reply-To: <191c3a030904021424ta0f4adwfe805f9d9bda86e@mail.gmail.com> References: <49D51AEE.7010904@gmx.net> <49D52142.7040401@gmx.net> <191c3a030904021424ta0f4adwfe805f9d9bda86e@mail.gmail.com> Message-ID: <49D53204.3090701@gmx.net> I restart FS and initiate an incoming call (trunk is registered at the SIP provider). This is what I see on the console: . . . 2009-04-02 23:39:16 [DEBUG] mod_event_socket.c:2224 mod_event_socket_runtime() Socket up listening on 0.0.0.0:8021 2009-04-02 23:39:16 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding xxx.xxx.xxx.xxx/32 (allow) to list strict 2009-04-02 23:39:16 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding xx.xx.xxx.xx/32 (allow) to list domains 2009-04-02 23:39:16 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 62.65.128.62/32 (allow) to list domains 2009-04-02 23:39:48 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() Can't find user [anonymous at 62.65.128.62] You must define a domain called '62.65.128.62' in your directory and add a user with the id="anonymous" attribute and you must configure your device to use the proper domain in it's authentication credentials. Nothing else. Here is the registration info: Name Netvoip Scheme Digest Realm sip.netvoip.ch Username 071xxxxxxx Password yes From Contact Exten 071xxxxxxx To sip:071xxxxxxx at sip.netvoip.ch Proxy sip:sip.netvoip.ch Context public Expires 60 Freq 60 Ping 0 PingFreq 0 State REGED Status UP CallsIN 0 CallsOUT 0 Best regards Peter Anthony Minessale schrieb: > look at the debug log and see what happens? > > On Thu, Apr 2, 2009 at 3:34 PM, Peter P GMX > wrote: > > My ACL contains: > > > > > > So this should be fine, right? However it doesn't work. > > Best regards > Peter > > > Brian West schrieb: > > We use the true network ip the invite came from NOT the one in > the sip > > headers. Not very trust worth to do that you think? ;) > > > > So if your ACL is correctly setup to 62.65.128.62 it would let > them in > > please verify your ACL is correct... > > > > /b > > > > On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote: > > > >> Hello, > >> > >> I am using a SIP account from Netvoip CH. I try to receive > inbound call > >> from this SIP trunk. I discovered that, when they sent an > invite, the > >> IP-Adress of the to: is their own IP address. > >> There fore ACL doesn't work and FS asks for authorization, > which then > >> fails > >> > >> I receive the following message on the CLI: > >> 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 > sofia_reg_parse_auth() > >> Can't find user [anonymous at 62.65.128.62 > >] > >> You must define a domain called '62.65.128.62' in your > directory and add > >> a user with the id="anonymous" attribute > >> and you must configure your device to use the proper domain in it's > >> authentication credentials. > > > > Brian West > > brian at freeswitch.org > > > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > 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/ > > 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 red.rain.seven at gmail.com Thu Apr 2 15:26:38 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Thu, 2 Apr 2009 15:26:38 -0700 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <49D51B19.3050709@gcdf.pl> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> <49D51218.2080209@gcdf.pl> <2CD5CE30-E2E5-4D72-A587-19F92D862665@freeswitch.org> <49D51B19.3050709@gcdf.pl> Message-ID: <59ad9ca10904021526s539417b1g7f5081f927ccf77@mail.gmail.com> How do you load balance conference calls? Doesn't all the conference members have to be on the same freeswitch server? On Thu, Apr 2, 2009 at 1:07 PM, Szymon Olko wrote: > Brian West pisze: > > what kind of hardware? > > > I made testes on Pentium-M laptop with single core 1,6Hz. I did not write > those results, it was over 100 calls that was handle > good, I was just curios what will happen. Tomorrow I will make real testes. > My production works on 2 core P4 and I have there only > 35 agents CPU load is like 7% with 15% small peeks. > > All phones are sip or analog via sip gateways, PRI is currently still on > asterisk which is connected via sip. > > > /b > > > > On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: > > > >> I did not described it perfectly. I made agents, queues scenarios on > >> conferences. > >> This what I tested was for example 100 calls, so it's 200 channels, > >> and 100 conferences, 2 channels per conference, all are > >> unmuted. I did that just because it is my work scenario. > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/b70e45ab/attachment-0001.html From bipin at xbipin.com Thu Apr 2 22:59:57 2009 From: bipin at xbipin.com (xbipin) Date: Thu, 2 Apr 2009 22:59:57 -0700 (PDT) Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <49D4DC5E.4080506@xbipin.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> <49D4CDEB.1040201@xbipin.com> <27317F98-DDB5-49D3-93B8-F0B8949710FB@freeswitch.org> <49D4DC5E.4080506@xbipin.com> Message-ID: <22862459.post@talk.nabble.com> hi, any1 have any idea how what to sue in dialplan such that calls from a single id go to a specific gateway only with blind registration enabled, this is the only major issue im having. Regards, Bipin -- View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22862459.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Thu Apr 2 23:53:35 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 3 Apr 2009 17:53:35 +1100 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <22862459.post@talk.nabble.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> <49D4CDEB.1040201@xbipin.com> <27317F98-DDB5-49D3-93B8-F0B8949710FB@freeswitch.org> <49D4DC5E.4080506@xbipin.com> <22862459.post@talk.nabble.com> Message-ID: <20090403065335.GA5645@jdc.jasonjgw.net> xbipin wrote: > > any1 have any idea how what to sue in dialplan such that calls from a single > id go to a specific gateway only with blind registration enabled, this is > the only major issue im having. Perhaps you could match the source address in the dial plan and then bridge or redirect the call to the desired gateway. for example. I tested a similar example once and it did work. From solko at gcdf.pl Fri Apr 3 00:27:23 2009 From: solko at gcdf.pl (Szymon Olko) Date: Fri, 03 Apr 2009 09:27:23 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <59ad9ca10904021526s539417b1g7f5081f927ccf77@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> <49D51218.2080209@gcdf.pl> <2CD5CE30-E2E5-4D72-A587-19F92D862665@freeswitch.org> <49D51B19.3050709@gcdf.pl> <59ad9ca10904021526s539417b1g7f5081f927ccf77@mail.gmail.com> Message-ID: <49D5BA5B.5070104@gcdf.pl> Henry Huang pisze: > How do you load balance conference calls? Doesn't all the conference > members have to be on the same freeswitch server? > As I wrote I do not load balance them yet. I didn't investigate that but what comes to my mind is to setup 2 FS end register agents to one of them (load balance them), sip phones through proxy server. Then one separate FS for incoming calls and in that FS place my queue system. When incoming call needs to be connected to agent then right FS machine would be choosen. This just idea I believe that in time I will need something like that FS developers will give us some modules or other options. > On Thu, Apr 2, 2009 at 1:07 PM, Szymon Olko wrote: > > Brian West pisze: > > what kind of hardware? > > > I made testes on Pentium-M laptop with single core 1,6Hz. I did not > write those results, it was over 100 calls that was handle > good, I was just curios what will happen. Tomorrow I will make real > testes. My production works on 2 core P4 and I have there only > 35 agents CPU load is like 7% with 15% small peeks. > > All phones are sip or analog via sip gateways, PRI is currently > still on asterisk which is connected via sip. > > > /b > > > > On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: > > > >> I did not described it perfectly. I made agents, queues scenarios on > >> conferences. > >> This what I tested was for example 100 calls, so it's 200 channels, > >> and 100 conferences, 2 channels per conference, all are > >> unmuted. I did that just because it is my work scenario. > > > > Brian West > > brian at freeswitch.org > > > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Henry Huang > UniC Solution - Communication Unified > VoIP & Open Source software Consultant > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 elhodred at gmail.com Fri Apr 3 01:19:08 2009 From: elhodred at gmail.com (Alfonso Pinto) Date: Fri, 3 Apr 2009 10:19:08 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls [SOLVED] Message-ID: <8b3b7acc0904030119m264656denaf6b261a398fff27@mail.gmail.com> Hi, Updating asterisk to version 1.4.24 solved the problem. Thanks guys. Regards. 2009/4/2 Brian West : > Follow this > thread?http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html > /b > On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the ?call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666 at 1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666 at 3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" ;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" ;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: . > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: . > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: . > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > 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 > > Brian West > brian at freeswitch.org > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 elhodred at gmail.com Fri Apr 3 01:22:27 2009 From: elhodred at gmail.com (Alfonso Pinto) Date: Fri, 3 Apr 2009 10:22:27 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <20090402003533.GA9849@jdc.jasonjgw.net> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> <20090402003533.GA9849@jdc.jasonjgw.net> Message-ID: <8b3b7acc0904030122v4a7ab910j79a6730adea59754@mail.gmail.com> Thank you so much, gmane gives me correct results. Instead, trying to search the thread Brian emailed to me with site:lists.freeswitch.org doesn't give the correct response, thread doesn't appears. Regards 2009/4/2 Jason White : > Alfonso Pinto wrote: >> One question more, maybe a stupid one: How can I search the archives? > > http://www.gmane.org/ > > The searching tool they use, Xapian, tends to give good relevance ranking, at > least in my experience. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 solko at gcdf.pl Fri Apr 3 03:05:30 2009 From: solko at gcdf.pl (Szymon Olko) Date: Fri, 03 Apr 2009 12:05:30 +0200 Subject: [Freeswitch-users] Slow freeswitch shutdown Message-ID: <49D5DF6A.4010204@gcdf.pl> In last SVN trunk version i noticed that stopping of freeswitch takes much time. I have configuration installed with freeswitch. I added sip gateway to my asterisk instance. I don't use asterisk currently and my gateway definition is like that: Starting freeswitch and shutting it down for console with '...' brings following logs. 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() UN-Registering example.com 2009-04-03 11:58:03 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting for worker thread 2009-04-03 11:58:03 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() deleted gateway example.com 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() UN-Registering 429956 2009-04-03 11:58:33 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting for worker thread 2009-04-03 11:58:33 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() deleted gateway 429956 Asterisk was not run at all so it should not register to it, why it hangs to unregister it? From codecomplete at free.fr Fri Apr 3 03:07:36 2009 From: codecomplete at free.fr (Fred) Date: Fri, 03 Apr 2009 12:07:36 +0200 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt Message-ID: <7.0.1.0.2.20090403120452.024273d8@fredshack.com> Carlos Talbot > Is there an interest in running FreeSWITCH on OpenWRT? I recently managed to compile and run a version for a MIPs based router, the Planex MZK-W04NU. Great news :-) I'm interested in running FS on any of this type of small hardware. Ideally, it should have a USB port so I can connect Sangoma's U100 connector to handle one or two POTS lines. Would the FS port you did handle this USB VoIP gateway? Thanks. From andy at fabulous4.co.uk Fri Apr 3 03:49:14 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Fri, 3 Apr 2009 11:49:14 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: Message-ID: <844E4DA20AAD4AB3B123D3A0572CCB5C@wsandy> Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:40 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please try SVN trunk. /b On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote: Hi Brian, 1.03 Thanks Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/c6ca6d2e/attachment.html From yivzhenko at mksat.net Fri Apr 3 05:43:51 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko (WP)) Date: Fri, 3 Apr 2009 15:43:51 +0300 Subject: [Freeswitch-users] Use rates from lcr in nibblebill module In-Reply-To: References: <200904021322.05690.yivzhenko@mksat.net> Message-ID: <200904031543.52605.yivzhenko@mksat.net> Thanks for variables and explanation. Work fine! Now wait for nibblebill can hangup connection when balance hits 0.00 On Thursday 02 April 2009 15:37:28 Rupa Schomaker wrote: > Update the to the latest. I've added more channel vars: > > eg: > > after doing: > > > (not a real number) > > I get the following: > > variable_lcr_query_digits: [12148267722] > variable_lcr_query_profile: [0] > variable_lcr_query_expanded_digits: [12148267722, 1214826772, 121482677, > 12148267, 1214826, 121482, 12148, 1214, 121, 12, 1] > variable_lcr_route_1: > [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gate >way/teliax/12148267722] variable_lcr_rate_1: [0.01000] > variable_lcr_carrier_1: [teliax] > variable_lcr_codec_1: [PCMU] > variable_lcr_route_2: > [[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec_string=PCMU]sofia/ga >teway/vitelity/12148267722] variable_lcr_rate_2: [0.01440] > variable_lcr_carrier_2: [vitelity] > variable_lcr_codec_2: [PCMU] > variable_lcr_route_count: [2] > variable_lcr_auto_route: > [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gate >way/teliax/12148267722|[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec >_string=PCMU]sofia/gateway/vitelity/12148267722] variable_import: > [lcr_carrier,lcr_rate] > > which, I think is what you are asking for. If you know which route you are > going to use (eg: 1) then you can get it's rate by using lcr_rate_1. > > Alternatively, you can use the lcr_auto_route and then once the b-leg > connects, query the b-leg variable for lcr_carrier and lcr_rate to see > which one was actually used. > > You really can't use lcr_auto_route and set a single rate since each leg > can be rated differently (look at example above). > > Normally lcr is used for your own rates between you and your carrier. That > is independant of the rate table used for your customers. You can use lcr > to query both. First use lcr to query your own rates using a different > profile. This would return a *single* route if you setup your route table > right. Save the rate in a var to be used with nibble bill. Then use lcr > with your external rates so you can route according to your own cost with > your carrier(s). > > On Thu, Apr 2, 2009 at 5:22 AM, Yuriy Ivzhenko (WP) wrote: > > Hi, > > > > I want to use module lcr to find a best route and his rate , then make a > > call > > and bill on that rate with nibblebill module. > > > > lcr return variable "lcr_auto_route" that contains "[lcr_rate=xxx]" > > variable > > for new channel. > > To use nibblebill i need to set "nibble_rate" = "lcr_rate". > > > > What is best method to do that? > > Is there a way to do that with standard tools, without use external > > scripts? > > > > > > Thanks, > > Yuriy > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 Apr 3 06:14:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 3 Apr 2009 08:14:41 -0500 Subject: [Freeswitch-users] Slow freeswitch shutdown In-Reply-To: <49D5DF6A.4010204@gcdf.pl> References: <49D5DF6A.4010204@gcdf.pl> Message-ID: <191c3a030904030614i7222eac2k9187c24d5d3e20e3@mail.gmail.com> update again and see if it's better On Fri, Apr 3, 2009 at 5:05 AM, Szymon Olko wrote: > In last SVN trunk version i noticed that stopping of freeswitch takes much > time. > > I have configuration installed with freeswitch. I added sip gateway to my > asterisk instance. I don't use asterisk currently and my > gateway definition is like that: > > > > > > > > > Starting freeswitch and shutting it down for console with '...' brings > following logs. > > 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() > UN-Registering example.com > 2009-04-03 11:58:03 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting > for worker thread > 2009-04-03 11:58:03 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() > deleted gateway example.com > 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() > UN-Registering 429956 > 2009-04-03 11:58:33 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting > for worker thread > 2009-04-03 11:58:33 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() > deleted gateway 429956 > > Asterisk was not run at all so it should not register to it, why it hangs > to unregister 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/30e13bde/attachment.html From lele at windmill.it Fri Apr 3 06:20:13 2009 From: lele at windmill.it (Lele Forzani) Date: Fri, 03 Apr 2009 15:20:13 +0200 Subject: [Freeswitch-users] codecs initialization flags in endpoint modules Message-ID: <1238764813.23024.102.camel@rivendell.windmill.it> Hello, I've been experimenting with the use of mod_dahdi_codec and other ways to perform external transcoding for codecs, and came up with noticing that transcoding resources seemed to be used up twice what I expected. That is and 2x the number of call legs, ending up to two encoder and two decoder instances per leg. So, I looked at the code and noticed almost every endpoint module does something like this (excerpt from mod_sofia, sofia_glue.c:~1800): if (switch_core_codec_init(&tech_pvt->read_codec, tech_pvt->iananame, tech_pvt->rm_fmtp, tech_pvt->rm_rate, tech_pvt->codec_ms, 1, SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | tech_pvt->profile->codec_flags, NULL, switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load codec?\n"); switch_goto_status(SWITCH_STATUS_FALSE, end); } if (switch_core_codec_init(&tech_pvt->write_codec, tech_pvt->iananame, tech_pvt->rm_fmtp, tech_pvt->rm_rate, tech_pvt->codec_ms, 1, SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | tech_pvt->profile->codec_flags, NULL, switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load codec?\n"); switch_goto_status(SWITCH_STATUS_FALSE, end); } The flags being SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE seems to be causing the apparent 'double' allocation of transcoding resources, and I fail to understand the need for both, in both cases. Could someone please spend a minute to explain? thanks lele From pablosaro at gmail.com Fri Apr 3 06:39:45 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 3 Apr 2009 10:39:45 -0300 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> Message-ID: <247f8100904030639l1f076cdt2f0f53303a236cc8@mail.gmail.com> Hi Ashley, A very simple HA solution can be achieved by using SRV. But according to your email, the solution that comes to my mind is the following: PSTN Gw --> OpenSIPs stateless w/ dispatcher module --> many FS boxes And if you want a balanced distribution of the calls, you can write a piece of code to keep statistics of your active sessions in a db. Each time a call arrives to a FS box, you trigger your piece of code to store a session record in a db and when the call ends you update the statistics in the db. This way, OpenSIPs can ask this db before making the decision where to route an incoming call. Fail over? If OpenSIPs gets a time out, just try with the next FS box. I hope it helps you. Pablo 2009/4/2 Ashley van Gerven > Hi, > > I can't find much info on setting up a redundant or heavy load FreeSwitch > implementation. Are there any > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? > > I imagine the entry level solution is to have two FS boxes configured > identitcally, with > redundant SBC software (recommendations?) in front, passing the calls to > the primary FS box, > or the backup FS box if the primary is not responding. Is that the easiest > solution? > > What about a situation of having a level of concurrent calls beyond what > one FS box can handle? I realise > that would be a very large number of concurrent calls, but we would need a > good plan on how to scale the > systems. > > Are there recommendations for load balancing solutions? Either soft or > hardware? > > My guess would be having 3 + 1 spare FS servers would work, where calls are > distributed accross 3 FS boxes > by a load balancer with one spare in event of failure. > > Also how would a FS box at max capacity behave? Does FS monitor available > resources and reject the > excess calls that it can't handle? Or would the load balancer have to be > configured with the maximum number > of calls per box? > > Would love to hear some experiences of deploying FS with failover & high > load. > > > Thanks > Ash > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/4bb02c9c/attachment.html From freeswitch-users at digitaldan.com Fri Apr 3 06:51:35 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Fri, 3 Apr 2009 07:51:35 -0600 (MDT) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <2659508.3491238766312290.JavaMail.daniel@radio> Message-ID: <24754670.3511238766668728.JavaMail.daniel@radio> I have my burst rate set to something low, 4096 right now. I also wrote a flash/flex app that has the same size buffer which results in the audio being heard immediately when connecting. As far as the audio being real time, the audio stream is about 6 seconds behind which I'm guessing is the result of the size of the lame buffers in the mod_shout modules (i'm using g.711 ulaw), I was going to look into that next week. Anyone have any thoughts about where else the delay may be happening? I hoping to get this down to around 2 seconds. D- ----- Original Message ----- From: "Rupa Schomaker" To: freeswitch-users at lists.freeswitch.org Sent: Thursday, April 2, 2009 8:05:38 AM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk 2009/4/2 Anthony Minessale < anthony.minessale at gmail.com > Its the buffering and startup of the shout stream taking up the time, HINT put the shoutcast stream into a local stream with a .loc file and then play that in the conference. Ah, that is easy enough! Though I think with icecast doing the burst_on_connect thingie there should be enough data (pushed much faster than real time) to fill FS's buffers. But that would require mod_shout to cooperate with that strategy. ie: on connect, drain the socket as fast as it can filling it's own buffers. Once it's own buffers are full start streaming. I think right now it drains the socket only as fast as it needs to. Or maybe not. I'll go the local stream route for now.... -- -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/20090403/f0de32fa/attachment.html From stormin.normin at hotmail.co.uk Fri Apr 3 07:02:45 2009 From: stormin.normin at hotmail.co.uk (Stromin Normin) Date: Fri, 3 Apr 2009 15:02:45 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: Thanks for all your help, I finally resolved the issue by setting comfort-noise to false in the conference.conf.xml. From: stormin.normin at hotmail.co.uk To: freeswitch-users at lists.freeswitch.org Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now _________________________________________________________________ Share your photos with Windows Live Photos ? Free. http://clk.atdmt.com/UKM/go/134665338/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/1218688b/attachment.html From brian at freeswitch.org Fri Apr 3 07:11:59 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Apr 2009 09:11:59 -0500 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: Did it sound more like a machine gun? /b On Apr 3, 2009, at 9:02 AM, Stromin Normin wrote: > Thanks for all your help, I finally resolved the issue by setting > comfort-noise to false in the conference.conf.xml. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/ff34b814/attachment.html From dujinfang at gmail.com Fri Apr 3 08:58:41 2009 From: dujinfang at gmail.com (dujinfang) Date: Fri, 3 Apr 2009 23:58:41 +0800 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? Message-ID: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> Hi, I have outbound gateways returns 403 or 503 constantly. So I tried to dial out using sofia/gateways/gw1/xxxx|sofia/gateways/gw2/xxxx|sofia/gateways/gw3... for fail over. To make it work, I need to set ignore_early_media=true. However, the caller do need to hear the early media to figure out what's going on. If I set ignore_early_media=false, only the first one tried. A little more detail: The gateway is first tier, if it cannot initiate a PSTN channel returns 403/503 immediately. If it can find a PSTN channel, but the callee fails, no answer or busy or others, it plays early media and returns 503. If I want failover, and the early media, how to do that? Thanks. regards, Seven. From msc at freeswitch.org Fri Apr 3 09:28:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 Apr 2009 09:28:59 -0700 Subject: [Freeswitch-users] codecs initialization flags in endpoint modules In-Reply-To: <1238764813.23024.102.camel@rivendell.windmill.it> References: <1238764813.23024.102.camel@rivendell.windmill.it> Message-ID: <87f2f3b90904030928t46fd697auacdc7d5ad01945a7@mail.gmail.com> FYI, these are good questions but they probably belong on the dev list since they are so technical in nature. :) -MC On Fri, Apr 3, 2009 at 6:20 AM, Lele Forzani wrote: > > Hello, > I've been experimenting with the use of mod_dahdi_codec and other ways > to perform external transcoding for codecs, and came up with noticing > that transcoding resources seemed to be used up twice what I expected. > That is and 2x the number of call legs, ending up to two encoder and two > decoder instances per leg. > > > So, I looked at the code and noticed almost every endpoint module does > something like this (excerpt from mod_sofia, sofia_glue.c:~1800): > > if (switch_core_codec_init(&tech_pvt->read_codec, > tech_pvt->iananame, > tech_pvt->rm_fmtp, > tech_pvt->rm_rate, > tech_pvt->codec_ms, > 1, > SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | > tech_pvt->profile->codec_flags, > NULL, switch_core_session_get_pool(tech_pvt->session)) != > SWITCH_STATUS_SUCCESS) { > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load > codec?\n"); > switch_goto_status(SWITCH_STATUS_FALSE, end); > } > > if (switch_core_codec_init(&tech_pvt->write_codec, > tech_pvt->iananame, > tech_pvt->rm_fmtp, > tech_pvt->rm_rate, > tech_pvt->codec_ms, > 1, > SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | > tech_pvt->profile->codec_flags, > NULL, switch_core_session_get_pool(tech_pvt->session)) != > SWITCH_STATUS_SUCCESS) { > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load > codec?\n"); > switch_goto_status(SWITCH_STATUS_FALSE, end); > } > > > The flags being SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE > seems to be causing the apparent 'double' allocation of transcoding > resources, and I fail to understand the need for both, in both cases. > > Could someone please spend a minute to explain? > > > thanks > lele > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/7c1f8511/attachment-0001.html From msc at freeswitch.org Fri Apr 3 09:30:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 Apr 2009 09:30:24 -0700 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: <87f2f3b90904030930r2b82a5c3oa9c558b4c5f7052e@mail.gmail.com> On Fri, Apr 3, 2009 at 7:11 AM, Brian West wrote: > Did it sound more like a machine gun? > /b > > Comfort noise for General Douglas McArthur I guess... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/ad368972/attachment.html From brian at freeswitch.org Fri Apr 3 10:04:57 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Apr 2009 12:04:57 -0500 Subject: [Freeswitch-users] MeucciSolutions@66.96.218.5 Message-ID: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> Does anyone else seem to be getting tons of calls from this evil IP? They keep ringing me via SIP over and over again. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/e692a43d/attachment.html From chris.chen2004 at gmail.com Fri Apr 3 10:36:18 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 3 Apr 2009 13:36:18 -0400 Subject: [Freeswitch-users] MeucciSolutions@66.96.218.5 In-Reply-To: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> References: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> Message-ID: <507898380904031036h546a2dc0x39d5927aac431830@mail.gmail.com> Hi Brian, looks like this Evil is calling everywhere today on port 5060, please see my asterisk log [Apr 3 11:13:42] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as05dbf888 [Apr 3 11:25:12] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as5ab1ec7b [Apr 3 11:25:44] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as5ab1ec7b [Apr 3 11:36:17] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as5c4625af [Apr 3 11:55:22] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as4d32ad06 [Apr 3 11:55:54] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as4d32ad06 [Apr 3 11:55:56] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as324c491b [Apr 3 12:00:19] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as4ab90c05 [Apr 3 12:14:43] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as3edfecbb [Apr 3 12:23:38] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as305dbb2e [Apr 3 12:32:14] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as5bf0ab42 [Apr 3 12:49:12] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as7f56ad67 [Apr 3 12:52:21] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as0d5d32e0 [Apr 3 13:10:09] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as1b806860 [Apr 3 13:17:46] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as487f8ecb [Apr 3 13:29:56] NOTICE[16920] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as613a9814 On Fri, Apr 3, 2009 at 1:04 PM, Brian West wrote: > Does anyone else seem to be getting tons of calls from this evil IP? They > keep ringing me via SIP over and over again. > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/2608f6d1/attachment.html From gkuri at ieee.org Fri Apr 3 10:53:53 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 03 Apr 2009 10:53:53 -0700 Subject: [Freeswitch-users] MeucciSolutions@66.96.218.5 In-Reply-To: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> References: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> Message-ID: <49D64D31.2060904@ieee.org> I heard about this a few days ago, they claim it's not them, but someone trying to "harm their reputation" ... http://www.meucci-solutions.com/complaints.asp?id=1 Gabe Brian West wrote: > Does anyone else seem to be getting tons of calls from this evil IP? > They keep ringing me via SIP over and over again. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 chris.chen2004 at gmail.com Fri Apr 3 11:02:15 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 3 Apr 2009 14:02:15 -0400 Subject: [Freeswitch-users] MeucciSolutions@66.96.218.5 In-Reply-To: <49D64D31.2060904@ieee.org> References: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> <49D64D31.2060904@ieee.org> Message-ID: <507898380904031102ieb3d6e4p9228fa728a5b2e1@mail.gmail.com> It is strange this IP is from US 66.96.218.5USUNITED STATESPENNSYLVANIASCRANTONNETWORK OPERATIONS CENTER INC On Fri, Apr 3, 2009 at 1:53 PM, Gabriel Kuri wrote: > I heard about this a few days ago, they claim it's not them, but someone > trying to "harm their reputation" ... > > http://www.meucci-solutions.com/complaints.asp?id=1 > > Gabe > > Brian West wrote: > > Does anyone else seem to be getting tons of calls from this evil IP? > > They keep ringing me via SIP over and over again. > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/86670c84/attachment.html From brian at freeswitch.org Fri Apr 3 11:09:55 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Apr 2009 13:09:55 -0500 Subject: [Freeswitch-users] MeucciSolutions@66.96.218.5 In-Reply-To: <507898380904031102ieb3d6e4p9228fa728a5b2e1@mail.gmail.com> References: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> <49D64D31.2060904@ieee.org> <507898380904031102ieb3d6e4p9228fa728a5b2e1@mail.gmail.com> Message-ID: <9F8ABE86-EE2A-4671-BFEE-E60A78047D76@freeswitch.org> Yes I opened a ticket with them about it... they said it would take 24 hours to figure anything out! /b On Apr 3, 2009, at 1:02 PM, Chris Chen wrote: > It is strange this IP is from US > 66.96.218.5 US UNITED STATES PENNSYLVANIA SCRANTON NETWORK > OPERATIONS CENTER INC > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/dd69580d/attachment-0001.html From carlos.talbot at gmail.com Fri Apr 3 14:29:01 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Fri, 3 Apr 2009 16:29:01 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <7.0.1.0.2.20090403120452.024273d8@fredshack.com> References: <7.0.1.0.2.20090403120452.024273d8@fredshack.com> Message-ID: <5800526b0904031429s3b1deb4do13ecf3335e18949a@mail.gmail.com> This would be ideal. I'm not sure though if the wanpipe kernel driver has been ported to openwrt (or non-x86 hardware for that matter). FYI, I'm slowly working on the wiki and have faced some obstacles as openwrt.org decided to upgrade their servers this past week and have been offline for a good part of that... Carlos On Fri, Apr 3, 2009 at 5:07 AM, Fred wrote: > Carlos Talbot > Is there an interest in running FreeSWITCH on > OpenWRT? I recently managed to compile and run a version for a MIPs > based router, the Planex MZK-W04NU. > > Great news :-) I'm interested in running FS on any of this type of > small hardware. Ideally, it should have a USB port so I can connect > Sangoma's U100 connector to handle one or two POTS lines. > > Would the FS port you did handle this USB VoIP gateway? > > 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/20090403/29d60877/attachment.html From kristian.kielhofner at gmail.com Fri Apr 3 14:42:29 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 3 Apr 2009 17:42:29 -0400 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> Message-ID: <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> You could try (although it's somewhat bleeding edge) to use OpenSIPS 1.5 with load_balancer (not heavily tested, btw) in front of some FreeSWITCH machines: http://www.opensips.org/html/docs/modules/devel/load_balancer.html 2009/4/2 Ashley van Gerven : > Hi, > > I can't find much info on setting up a redundant or heavy load FreeSwitch > implementation. Are there any > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? > > I imagine the entry level solution is to have two FS boxes configured > identitcally, with > redundant SBC software (recommendations?) in front, passing the calls to the > primary FS box, > or the backup FS box if the primary is not responding. Is that the easiest > solution? > > What about a situation of having a level of concurrent calls beyond what one > FS box can handle? I realise > that would be a very large number of concurrent calls, but we would need a > good plan on how to scale the > systems. > > Are there recommendations for load balancing solutions? Either soft or > hardware? > > My guess would be having 3 + 1 spare FS servers would work, where calls are > distributed accross 3 FS boxes > by a load balancer with one spare in event of failure. > > Also how would a FS box at max capacity behave? Does FS monitor available > resources and reject the > excess calls that it can't handle? Or would the load balancer have to be > configured with the maximum number > of calls per box? > > Would love to hear some experiences of deploying FS with failover & high > load. > > > Thanks > Ash > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From pablosaro at gmail.com Fri Apr 3 15:30:24 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 3 Apr 2009 19:30:24 -0300 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> Message-ID: <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> Hi Kristian, you're right. Definitively that will be best solution as soon as it's released as stable (it's alpha now). http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing Pablo On Fri, Apr 3, 2009 at 6:42 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > You could try (although it's somewhat bleeding edge) to use OpenSIPS > 1.5 with load_balancer (not heavily tested, btw) in front of some > FreeSWITCH machines: > > http://www.opensips.org/html/docs/modules/devel/load_balancer.html > > 2009/4/2 Ashley van Gerven : > > Hi, > > > > I can't find much info on setting up a redundant or heavy load FreeSwitch > > implementation. Are there any > > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment? > > > > I imagine the entry level solution is to have two FS boxes configured > > identitcally, with > > redundant SBC software (recommendations?) in front, passing the calls to > the > > primary FS box, > > or the backup FS box if the primary is not responding. Is that the > easiest > > solution? > > > > What about a situation of having a level of concurrent calls beyond what > one > > FS box can handle? I realise > > that would be a very large number of concurrent calls, but we would need > a > > good plan on how to scale the > > systems. > > > > Are there recommendations for load balancing solutions? Either soft or > > hardware? > > > > My guess would be having 3 + 1 spare FS servers would work, where calls > are > > distributed accross 3 FS boxes > > by a load balancer with one spare in event of failure. > > > > Also how would a FS box at max capacity behave? Does FS monitor available > > resources and reject the > > excess calls that it can't handle? Or would the load balancer have to be > > configured with the maximum number > > of calls per box? > > > > Would love to hear some experiences of deploying FS with failover & high > > load. > > > > > > Thanks > > Ash > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/4e87c223/attachment.html From grevenx at me.com Fri Apr 3 15:48:00 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Sat, 04 Apr 2009 00:48:00 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> Message-ID: Where do you guys read that it's in alpha? On the opensips.org they proclaim OpenSips 1.5 released, with that module being one of the new features. I don't see any mention of it being alpha/beta functionality? Best regards, Even Andr? On 4. april. 2009, at 00.30, Pablo Hernan Saro wrote: > Hi Kristian, you're right. Definitively that will be best solution > as soon as it's released as stable (it's alpha now). > http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing > > Pablo > > On Fri, Apr 3, 2009 at 6:42 PM, Kristian Kielhofner > wrote: > You could try (although it's somewhat bleeding edge) to use OpenSIPS > 1.5 with load_balancer (not heavily tested, btw) in front of some > FreeSWITCH machines: > > http://www.opensips.org/html/docs/modules/devel/load_balancer.html > > 2009/4/2 Ashley van Gerven : > > Hi, > > > > I can't find much info on setting up a redundant or heavy load > FreeSwitch > > implementation. Are there any > > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment > ? > > > > I imagine the entry level solution is to have two FS boxes > configured > > identitcally, with > > redundant SBC software (recommendations?) in front, passing the > calls to the > > primary FS box, > > or the backup FS box if the primary is not responding. Is that the > easiest > > solution? > > > > What about a situation of having a level of concurrent calls > beyond what one > > FS box can handle? I realise > > that would be a very large number of concurrent calls, but we > would need a > > good plan on how to scale the > > systems. > > > > Are there recommendations for load balancing solutions? Either > soft or > > hardware? > > > > My guess would be having 3 + 1 spare FS servers would work, where > calls are > > distributed accross 3 FS boxes > > by a load balancer with one spare in event of failure. > > > > Also how would a FS box at max capacity behave? Does FS monitor > available > > resources and reject the > > excess calls that it can't handle? Or would the load balancer have > to be > > configured with the maximum number > > of calls per box? > > > > Would love to hear some experiences of deploying FS with failover > & high > > load. > > > > > > Thanks > > Ash > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090404/36d7c009/attachment.html From pablosaro at gmail.com Fri Apr 3 16:24:58 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 3 Apr 2009 20:24:58 -0300 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> Message-ID: <247f8100904031624s4a4ea40v4a5c0fd6edd71b42@mail.gmail.com> Not opensips but the module is in alpha. In the modules doc page says "alpha/new". Pablo On 4/3/09, Even Andr? Fiskvik wrote: > Where do you guys read that it's in alpha? > On the opensips.org they proclaim OpenSips 1.5 released, > with that module being one of the new features. I don't see any > mention of it being alpha/beta functionality? > > Best regards, > Even Andr? > > On 4. april. 2009, at 00.30, Pablo Hernan Saro wrote: > >> Hi Kristian, you're right. Definitively that will be best solution >> as soon as it's released as stable (it's alpha now). >> http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing >> >> Pablo >> >> On Fri, Apr 3, 2009 at 6:42 PM, Kristian Kielhofner >> > > wrote: >> You could try (although it's somewhat bleeding edge) to use OpenSIPS >> 1.5 with load_balancer (not heavily tested, btw) in front of some >> FreeSWITCH machines: >> >> http://www.opensips.org/html/docs/modules/devel/load_balancer.html >> >> 2009/4/2 Ashley van Gerven : >> > Hi, >> > >> > I can't find much info on setting up a redundant or heavy load >> FreeSwitch >> > implementation. Are there any >> > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment >> ? >> > >> > I imagine the entry level solution is to have two FS boxes >> configured >> > identitcally, with >> > redundant SBC software (recommendations?) in front, passing the >> calls to the >> > primary FS box, >> > or the backup FS box if the primary is not responding. Is that the >> easiest >> > solution? >> > >> > What about a situation of having a level of concurrent calls >> beyond what one >> > FS box can handle? I realise >> > that would be a very large number of concurrent calls, but we >> would need a >> > good plan on how to scale the >> > systems. >> > >> > Are there recommendations for load balancing solutions? Either >> soft or >> > hardware? >> > >> > My guess would be having 3 + 1 spare FS servers would work, where >> calls are >> > distributed accross 3 FS boxes >> > by a load balancer with one spare in event of failure. >> > >> > Also how would a FS box at max capacity behave? Does FS monitor >> available >> > resources and reject the >> > excess calls that it can't handle? Or would the load balancer have >> to be >> > configured with the maximum number >> > of calls per box? >> > >> > Would love to hear some experiences of deploying FS with failover >> & high >> > load. >> > >> > >> > Thanks >> > Ash >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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://blog.krisk.org >> http://www.submityoursip.com >> http://www.astlinux.org >> http://www.star2star.com >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 Gmail for mobile | mobile.google.com From jason at jasonjgw.net Fri Apr 3 16:53:00 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 4 Apr 2009 10:53:00 +1100 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> Message-ID: <20090403235300.GA10045@jdc.jasonjgw.net> dujinfang wrote: > However, the caller do need to hear the early media to figure out > what's going on. If I set ignore_early_media=false, only the first one > tried. Could you use ring_ready? that way, the calling SIP phone should generate the ringback. From brian at freeswitch.org Fri Apr 3 17:13:57 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Apr 2009 19:13:57 -0500 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <20090403235300.GA10045@jdc.jasonjgw.net> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> Message-ID: <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> First one to give media wins unless you ignore_early_media /b On Apr 3, 2009, at 6:53 PM, Jason White wrote: > Could you use ring_ready? that way, the calling SIP phone should > generate the > ringback. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/e7881c46/attachment.html From kristian.kielhofner at gmail.com Fri Apr 3 17:45:27 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 3 Apr 2009 20:45:27 -0400 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> Message-ID: <2d9149cd0904031745r4f5bc194hc7718159c1875b0d@mail.gmail.com> Pablo, It is very cool and a very compelling reason to upgrade/move to OpenSIPS 1.5. I'm running (mostly) OpenSIPS 1.4.4/1.4.5 now and it's rock solid (as usual). It's really an excellent complement to FreeSWITCH! I will be doing testing with 1.5 and the new load balancer module shortly. On Fri, Apr 3, 2009 at 6:30 PM, Pablo Hernan Saro wrote: > Hi Kristian, you're right. Definitively that will be best solution as soon > as it's released as stable (it's alpha now). > http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing > > Pablo > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From pablosaro at gmail.com Fri Apr 3 20:48:06 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Sat, 4 Apr 2009 00:48:06 -0300 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <2d9149cd0904031745r4f5bc194hc7718159c1875b0d@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> <2d9149cd0904031745r4f5bc194hc7718159c1875b0d@mail.gmail.com> Message-ID: <247f8100904032048v316454b5rceb685b816d88477@mail.gmail.com> Hi Kristian, Let us know your experience as soon as you try it. Why not write a wiki page? =) On Fri, Apr 3, 2009 at 9:45 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Pablo, > > It is very cool and a very compelling reason to upgrade/move to > OpenSIPS 1.5. I'm running (mostly) OpenSIPS 1.4.4/1.4.5 now and it's > rock solid (as usual). It's really an excellent complement to > FreeSWITCH! > > I will be doing testing with 1.5 and the new load balancer module shortly. > > On Fri, Apr 3, 2009 at 6:30 PM, Pablo Hernan Saro > wrote: > > Hi Kristian, you're right. Definitively that will be best solution as > soon > > as it's released as stable (it's alpha now). > > http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing > > > > Pablo > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090404/42375ca4/attachment.html From zhaoxxqq at 163.com Fri Apr 3 22:33:07 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Sat, 4 Apr 2009 13:33:07 +0800 Subject: [Freeswitch-users] compile problem in vista. Message-ID: <200904041333057523168@163.com> Hi, It's first time I install FS in Vista. After having downloaded the FS sources from svn. Follow the instruction on how to build FS on Windows. I Using Visual C++ 2008 Express Open Freeswitch.sln Right click the main solution node at the top of the Solution Explorer Right click and select Build after do this I was stoped by the problem. the error is like below, what need I to do? anyone can help me? Error 6 error C2008: '#' : unexpected in macro definition c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.h 1532 Error 8 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 9 error C2065: 'defiTE_a_15' : undeclared identifier c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 10 error C2099: initializer is not a constant c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 11 error C2061: syntax error : identifier 'defiTE_a_15' c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 12 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 13 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 14 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 15 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 16 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 17 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 18 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 19 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 20 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 21 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 22 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 23 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 24 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal.c': No such file or directory c1 Error 25 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_diphone.c': No such file or directory c1 Error 26 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_lpc.c': No such file or directory c1 Error 27 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_res.c': No such file or directory c1 Error 28 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_residx.c': No such file or directory c1 Error 30 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_aswd.c': No such file or directory c1 Error 31 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_dur_stats.c': No such file or directory c1 Error 32 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_durz_cart.c': No such file or directory c1 Error 33 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_expand.c': No such file or directory c1 Error 34 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_f0_model.c': No such file or directory c1 Error 35 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_f0lr.c': No such file or directory c1 Error 36 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_ffeatures.c': No such file or directory c1 Error 37 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_gpos.c': No such file or directory c1 Error 38 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_int_accent_cart.c': No such file or directory c1 Error 39 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_int_tone_cart.c': No such file or directory c1 Error 40 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_nums_cart.c': No such file or directory c1 Error 41 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_phoneset.c': No such file or directory c1 Error 42 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_phrasing_cart.c': No such file or directory c1 Error 43 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_text.c': No such file or directory c1 Error 44 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\usenglish.c': No such file or directory c1 Error 47 fatal error C1083: Cannot open include file: 'usenglish.h': No such file or directory c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt.c 46 Error 103 fatal error C1083: Cannot open include file: 'usenglish.h': No such file or directory c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb.c 46 Error 109 fatal error C1083: Cannot open include file: 'usenglish.h': No such file or directory c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms.c 46 Error 123 error C2220: warning treated as error - no 'object' file generated c:\Users\lenovo\Documents\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua.c 1 Error 140 error C2695: 'LUA::Session::destroy': overriding virtual function differs from 'CoreSession::destroy' only by calling convention c:\users\lenovo\documents\freeswitch\src\mod\languages\mod_lua\freeswitch_lua.h 26 Error 149 fatal error LNK1181: cannot open input file 'flite.lib' mod_flite Error 175 fatal error LNK1181: cannot open input file '..\..\..\..\libs\win32\sofia\debug\libsofia_sip_ua_static.lib' mod_sofia 2009-04-04 zhaoxxqq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090404/87880a4e/attachment-0001.html From dujinfang at gmail.com Fri Apr 3 23:35:18 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 4 Apr 2009 14:35:18 +0800 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <20090403235300.GA10045@jdc.jasonjgw.net> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> Message-ID: On Apr 4, 2009, at 7:53 AM, Jason White wrote: > dujinfang wrote: >> However, the caller do need to hear the early media to figure out >> what's going on. If I set ignore_early_media=false, only the first >> one >> tried. > > Could you use ring_ready? that way, the calling SIP phone should > generate the > ringback. > ring_ready would be replaced by remote party early media. It does not harm though, I still need to listen early media. 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 dujinfang at gmail.com Fri Apr 3 23:41:03 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 4 Apr 2009 14:41:03 +0800 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> Message-ID: On Apr 4, 2009, at 8:13 AM, Brian West wrote: > First one to give media wins unless you ignore_early_media > > /b > Thanks, I tested again. That's exactly what I want except the problem sometimes the gateway gives (wrong)early_media but fails immediately, so I have no chance to hear the early media. And unfortunately the gateway is beyond my control. :( Will do more testing. > > On Apr 3, 2009, at 6:53 PM, Jason White wrote: > >> Could you use ring_ready? that way, the calling SIP phone should >> generate the >> ringback. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090404/cec0e888/attachment.html From kristian.kielhofner at gmail.com Sat Apr 4 00:13:44 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Sat, 4 Apr 2009 03:13:44 -0400 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <247f8100904032048v316454b5rceb685b816d88477@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> <2d9149cd0904031745r4f5bc194hc7718159c1875b0d@mail.gmail.com> <247f8100904032048v316454b5rceb685b816d88477@mail.gmail.com> Message-ID: <2d9149cd0904040013u180eed38q4c1ff09dd8587487@mail.gmail.com> Hey Pablo, Wiki page? I just might! :) On Fri, Apr 3, 2009 at 11:48 PM, Pablo Hernan Saro wrote: > Hi Kristian, > > Let us know your experience as soon as you try it. Why not write a wiki > page?? =) > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Sat Apr 4 00:32:38 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Sat, 4 Apr 2009 03:32:38 -0400 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> Message-ID: <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> On Sat, Apr 4, 2009 at 2:41 AM, dujinfang wrote: > > On Apr 4, 2009, at 8:13 AM, Brian West wrote: > > First one to give media wins unless you ignore_early_media > /b > > Thanks, I tested again. That's exactly what I want except the problem > sometimes the gateway gives (wrong)early_media but fails immediately, so I > have no chance to hear the early media. And unfortunately the gateway is > beyond my control. :( > Will do more testing. > I'm not really sure how else FS should handle this... As Brian said "first one with media wins". That's the problem with early media. Is it ringback that might turn into a completed call? Is it some error message played to the user? Is it someones voicemail system, trying to save some money? One way or another, is it something the user should hear? No way to know, really... 180/183 with SDP is a bit ambiguous. I always get frustrated when various people /insist/ on using 183 w/ SDP just for ringback. Have they never heard of 180 w/o SDP? Let me generate my own local ringback and/or handle the situation accordingly! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From dave at 3c.co.uk Sun Apr 5 20:12:17 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 06 Apr 2009 04:12:17 +0100 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> Message-ID: <1238987537.13971.13.camel@dk-d820> On Sat, 2009-04-04 at 03:32 -0400, Kristian Kielhofner wrote: > > 180/183 with SDP is a bit ambiguous. I always get frustrated when > various people /insist/ on using 183 w/ SDP just for ringback. Have > they never heard of 180 w/o SDP? Let me generate my own local > ringback and/or handle the situation accordingly! Ah, well, that's where you're trying to change the way that things have been done for some decades. Ringback has historically been generated close to the called party, which is why you hear different ringback if you call people in different countries. Furthermore, that audio path is used to convey all sorts of messages: "the number you have called has been changed", "the cellphone you have called has not responded", "calls to 1-800 numbers are not free if made from overseas.." Lastly, there's no guarantee that it'll be possible to differentiate between one of these and ringback from the signalling alone and, in many cases, there is simply no mechanism available to provide such differentiation. You're probably best advised to swim with the tide on this one..! Cheers -- Dave From brian at freeswitch.org Sun Apr 5 20:29:34 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 5 Apr 2009 22:29:34 -0500 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <1238987537.13971.13.camel@dk-d820> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> Message-ID: <2DCE0758-C4F4-4EEB-9808-DBA7344DF48B@freeswitch.org> Yes there was till the SIP RFC writers happen to make their ears rather sore! (RCI) 180 vs 183 should have been it... but NO they had to be ambiguous about that too... if you get a 180 without an SDP you generate... 180 or 183 with SDP (they had a sense of humor about this one I think!) Then this one tops the cake... on early media with forked dial... Say you call billy, mary and ken at the same time. Billy's address provides early media (ringing) you are to give the first one to respond with media to the caller... but if by chance Mary's phone provider is having a problem and they give congestion 20 seconds later and actually answer the call to do this cuz you know how stupid telco's are... now you are to give the caller the congestion tone... So you had prefect ringing.. then congestion... I think we have all be there, heard that! /b On Apr 5, 2009, at 10:12 PM, David Knell wrote: > signalling alone and, in many cases, there is simply no mechanism > available to provide such differentiation. From kristian.kielhofner at gmail.com Sun Apr 5 21:08:16 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 6 Apr 2009 00:08:16 -0400 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <1238987537.13971.13.camel@dk-d820> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> Message-ID: <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> On Sun, Apr 5, 2009 at 11:12 PM, David Knell wrote: > > Ah, well, that's where you're trying to change the way that things > have been done for some decades. ?Ringback has historically been > generated close to the called party, which is why you hear different > ringback if you call people in different countries. What's wrong with that? Isn't that what we are all doing (or trying to do), to some extent? International dialing very well may use different ringbacks but: 1) How important is this, really? 2) How much more complicated is adding at least the real potential for 180? Actually using 180 w/o SDP provides for enhanced call handing functionality while only requiring (in many cases) one additional test scenario. Consider the current example (all 180s are actually 180s w/o SDP and 183 is 183 w/ SDP): Bridging a call to multiple destinations (A, B, and C). A: 100,180 B: 100,180,200 C: 100,183 We could have implemented proper forking if it weren't for C who insisted on sending media early (for whatever reason). While I could see many scenarios where this might happen even with the configuration I suggest, consider what would happen in the ideal scenario: A: 100,180 B: 100,180,200 C: 100,180 Ah, B won because it was the first endpoint to actually /answer/ the call and begin playing media. Nice and clean. This is what happens when dialing local phones behind a PBX. All endpoint SIP phones send 180 to allow for clean parallel forking across them. This is what makes configuration for ring groups, etc, possible. I'm not suggesting that this configuration could be simply "dropped in" when dialing to the PSTN but it should at least be a a possibility. I suppose the other thing here (which is possible and has been suggested) is to configure your device to ignore early media. Too bad (due to various reasons, some of them being legacy PSTN) that in some cases the user should hear that 183. Speaking of which... > Furthermore, that audio path is used to convey all sorts of messages: > "the number you have called has been changed", "the cellphone you have > called has not responded", "calls to 1-800 numbers are not free if > made from overseas.." ?Lastly, there's no guarantee that it'll be > possible to differentiate between one of these and ringback from the > signalling alone and, in many cases, there is simply no mechanism > available to provide such differentiation. People poke at SIP all the time for this one but this is where the PSTN even seems a bit ambiguous. We have ISDN cause codes AND inband audio messages? I'm reminded of a situation the other day with a provider's SIP architecture. If you send a call to a completely bogus destination number (1, in this case) they reply with an inband audio error message. Why not send a 404 or something that is easily parsed and understood by my platform (FreeSWITCH)? In this case I needed to do some further action in the event of a "call failure" and as far as bridge/mod_sofia is concerned this was a "successful" call. I know this specific instance could be avoided but I can't wait to see what else they play inband audio messages for. Of course I can't really configure my end to ignore early media because I could miss out on some legit inband audio messaging that is actually useful. > You're probably best advised to swim with the tide on this one..! If I "swam with the tide" I'd probably be out getting my CCIE and installing Call Manager systems or something ;). Maybe that's not the best or the most "fair" analogy but hopefully you can see my point. I think there's a little rebel in all of us here on freeswitch-users! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From andy at fabulous4.co.uk Mon Apr 6 02:07:25 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 6 Apr 2009 10:07:25 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: Message-ID: Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:40 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please try SVN trunk. /b On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote: Hi Brian, 1.03 Thanks Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/dfc8dedd/attachment-0001.html From helmut.kuper at ewetel.de Mon Apr 6 03:20:25 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 06 Apr 2009 12:20:25 +0200 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> <49CB9E0C.4030300@ewetel.de> <49CCEC15.8010500@ewetel.de> Message-ID: <49D9D769.5020101@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I still have this problem. From the day of starting up freeswitch two threads are consuming slowly more and more CPU power. In parallel FS virtual and physical memory usage grows slowly as well. FS is up for 6 days now and served 3160 sessions. Virtual memory usage has grown from 200MB to 1.1GB (18,1%) and is still growing. CPU is mostly around 25% with lowest of 17% and a maximum of 50% (all on a 32 bit 4 CPU core system) and is still growing. There are two FS-Threads with nearly same CPU usage of currently around 20% each (I used htop for this): strace Thread 1 (I guess this is the sofia/sip thread): epoll_wait(21, {}, 4, 0) = 0 epoll_wait(21, {}, 4, 0) = 0 epoll_wait(21, {{EPOLLIN, {u32=2, u64=2}}}, 4, 0) = 1 ioctl(24, FIONREAD, [268]) = 0 recvmsg(24, {msg_name(16)={sa_family=AF_INET, sin_port=htons(1068), sin_addr=inet_addr("85.16.245.249")}, msg_iov(1)=[{"SIP/2.0 200 Ok\r\nVia: SIP/2.0/UDP"..., 268}], msg_controllen=0, msg_flags=0}, 0) = 268 gettimeofday({1239012809, 343580}, NULL) = 0 gettimeofday({1239012809, 343645}, NULL) = 0 epoll_wait(21, {}, 4, 0) = 0 epoll_wait(21, {}, 4, 0) = 0 ... strace Thread 2: select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) ... I use FreeSWITCH Version 1.0.trunk (12347M) regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFJ2ddp4tZeNddg3dwRAlkXAJ9fIwpJw6u18JPhFC4hzB+0Z1iAbgCfW7AE dnrmpXDLVOnWtjwFKMoVw48= =zzZ9 -----END PGP SIGNATURE----- -------------- next part -------------- A non-text attachment was scrubbed... Name: 0xD760DDDC.asc Type: application/pgp-keys Size: 1854 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/f512e936/attachment.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: 0xD760DDDC.asc Type: application/pgp-keys Size: 1854 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/f512e936/attachment-0001.bin From steveu at coppice.org Mon Apr 6 04:40:41 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 06 Apr 2009 19:40:41 +0800 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <2DCE0758-C4F4-4EEB-9808-DBA7344DF48B@freeswitch.org> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> <2DCE0758-C4F4-4EEB-9808-DBA7344DF48B@freeswitch.org> Message-ID: <49D9EA39.4010404@coppice.org> Brian West wrote: > Say you call billy, mary and ken at the same time. Billy's address > provides early media (ringing) you are to give the first one to > respond with media to the caller... but if by chance Mary's phone > provider is having a problem and they give congestion 20 seconds later > and actually answer the call to do this cuz you know how stupid > telco's are... now you are to give the caller the congestion tone... > So you had prefect ringing.. then congestion... I think we have all be > there, heard that! > Er, that's not stupidity. If the regulations allow them to answer at this point, they will. It generates revenue. Its a disaster for a lot of services which need to know if the call was answered to tell what to do next, but it ain't done through stupidity. We are the stupid suckers who pay. Steve From codecomplete at free.fr Mon Apr 6 04:41:59 2009 From: codecomplete at free.fr (Fred) Date: Mon, 06 Apr 2009 13:41:59 +0200 Subject: [Freeswitch-users] Freeswitch not started OK at boottime? Message-ID: <7.0.1.0.2.20090406133425.05092870@fredshack.com> Hello I'm having a problem connecting to the Freeswitch server running on a Suse server when the it's started at bootime, but OK if I start it manually through the init.d script, so I guess I did something wrong when setting things up. Here's what I did: 1. Downloaded and compiled the latest SVN source 2. cp /usr/src/freeswitch/build/freeswitch.init.suse /etc/init.d/freeswitch 3. chmod 755 /etc/init.d/freeswitch 4. chkconfig freeswitch 345 5. chkconfig -l freeswitch 6. (why needed in addition to chkconfig?) ln -s /etc/init.d/freeswitch /usr/sbin/rcfreeswitch 7. Edit /etc/init.d/freeswitch: FREESWITCH_BIN=/usr/local/freeswitch/bin/freeswitch #(BAD!) FREESWITCH_CONFIG=/usr/local/freeswitch/conf/freeswitch.xml FREESWITCH_PARAMS="-nc" Here's what it says when I try to connect to the server: ========= # ps aux | grep free root 3497 0.6 0.7 16912 8212 ? Sl 12:03 0:00 /usr/local/freeswitch/bin/freeswitch -nc # cd /usr/local/freeswitch/bin/ # ./fs_cli [ERROR] libs/esl/fs_cli.c:642 main() Error Connecting [Socket Connection Error] ========= Here's how to solve this issue manually: ========= # /etc/init.d/freeswitch stop Shutting down FreeSWITCH done # /etc/init.d/freeswitch start Starting FreeSWITCH 3867 Backgrounding. done /usr/local/freeswitch/bin # ./fs_cli [logo deleted] +OK log level [7] freeswitch at internal> /exit # ========= Any idea what is wrong? Thank you for any hint. From steveu at coppice.org Mon Apr 6 04:43:05 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 06 Apr 2009 19:43:05 +0800 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> Message-ID: <49D9EAC9.8090804@coppice.org> Kristian Kielhofner wrote: > On Sun, Apr 5, 2009 at 11:12 PM, David Knell wrote: > >> Ah, well, that's where you're trying to change the way that things >> have been done for some decades. Ringback has historically been >> generated close to the called party, which is why you hear different >> ringback if you call people in different countries. >> > > What's wrong with that? Isn't that what we are all doing (or trying > to do), to some extent? > > International dialing very well may use different ringbacks but: > > 1) How important is this, really? > 2) How much more complicated is adding at least the real potential for 180? > The actual ringback tone may not be important, but many other supervisory indications may occur at that point, either as tones or as voice announcements. E.g. call a cellphone that has problems - out of range, out of service, etc - and you will probably get a voice announcement telling you want's up. Steve From brian at freeswitch.org Mon Apr 6 06:24:05 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Apr 2009 08:24:05 -0500 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: References: Message-ID: Please update... rebootstrap.. you caught SVN with the libtool patch which kinda broken a few things linking. /b On Apr 6, 2009, at 4:07 AM, Andy Ayers wrote: > Hi Brian, > > I've upgraded to svn trunk but am now getting errors on load which > are preventing it from working: > > 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_shout.so > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: > ogg_stream_pagein** > 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_spidermonkey.so > **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: > PR_LocalTimeParameters** > > Sorry if this is obvious but what have I done wrong? > > Thanks for your help > Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/a1d4bc81/attachment.html From brian at freeswitch.org Mon Apr 6 06:31:41 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Apr 2009 08:31:41 -0500 Subject: [Freeswitch-users] Freeswitch not started OK at boottime? In-Reply-To: <7.0.1.0.2.20090406133425.05092870@fredshack.com> References: <7.0.1.0.2.20090406133425.05092870@fredshack.com> Message-ID: <0FEBBD98-0B4E-43F9-85D5-D7C66D290029@freeswitch.org> What run level are you starting freeswitch? /b On Apr 6, 2009, at 6:41 AM, Fred wrote: > Hello > > I'm having a problem connecting to the Freeswitch server running on a > Suse server when the it's started at bootime, but OK if I start it > manually through the init.d script, so I guess I did something wrong > when setting things up. > > Here's what I did: > 1. Downloaded and compiled the latest SVN source > 2. cp /usr/src/freeswitch/build/freeswitch.init.suse /etc/init.d/ > freeswitch > 3. chmod 755 /etc/init.d/freeswitch > 4. chkconfig freeswitch 345 > 5. chkconfig -l freeswitch > 6. (why needed in addition to chkconfig?) ln -s > /etc/init.d/freeswitch /usr/sbin/rcfreeswitch > 7. Edit /etc/init.d/freeswitch: > FREESWITCH_BIN=/usr/local/freeswitch/bin/freeswitch > #(BAD!) FREESWITCH_CONFIG=/usr/local/freeswitch/conf/freeswitch.xml > FREESWITCH_PARAMS="-nc" From dujinfang at gmail.com Mon Apr 6 06:46:42 2009 From: dujinfang at gmail.com (dujinfang) Date: Mon, 6 Apr 2009 21:46:42 +0800 Subject: [Freeswitch-users] Freeswitch not started OK at boottime? In-Reply-To: <7.0.1.0.2.20090406133425.05092870@fredshack.com> References: <7.0.1.0.2.20090406133425.05092870@fredshack.com> Message-ID: <16CBB37E-E274-4B14-9EAA-CEE1DC679A6B@gmail.com> > Here's what it says when I try to connect to the server: > ========= > # ps aux | grep free > root 3497 0.6 0.7 16912 8212 ? Sl 12:03 0:00 > /usr/local/freeswitch/bin/freeswitch -nc > It seems started, I never used a suse, however, can you try this? #netstat -an | grep 8021 Maybe FS started before network is ready. Check scripts in /etc/ rc.d/ or any equiv > # cd /usr/local/freeswitch/bin/ > # ./fs_cli > [ERROR] libs/esl/fs_cli.c:642 main() Error Connecting [Socket > Connection Error] > ========= From dave at 3c.co.uk Mon Apr 6 06:47:16 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 06 Apr 2009 14:47:16 +0100 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> Message-ID: <1239025636.12559.13.camel@dk-d820> On Mon, 2009-04-06 at 00:08 -0400, Kristian Kielhofner wrote: > Actually using 180 w/o SDP provides for enhanced call handing > functionality while only requiring (in many cases) one additional test > scenario. Consider the current example (all 180s are actually 180s > w/o SDP and 183 is 183 w/ SDP): > > Bridging a call to multiple destinations (A, B, and C). > > A: 100,180 > B: 100,180,200 > C: 100,183 > > We could have implemented proper forking if it weren't for C who > insisted on sending media early (for whatever reason). While I could > see many scenarios where this might happen even with the configuration > I suggest, consider what would happen in the ideal scenario: > > A: 100,180 > B: 100,180,200 > C: 100,180 > Ah, B won because it was the first endpoint to actually /answer/ the > call and begin playing media. Nice and clean. Hang on - if you want to bridge the call on *answer*, then bridge it on answer, not when one leg starts sending you early media. I've no idea if FS supports this behaviour for its forked dialling, but it's easy to do with a bunch of originates, and uuid_bridge the inbound leg to the first one which answers. > People poke at SIP all the time for this one but this is where the > PSTN even seems a bit ambiguous. We have ISDN cause codes AND inband > audio messages? Yes. A clearing code is used when the call's cleared; inband audio can be used to give the caller more information than a simple clearing code might allow - for example, "The number you are calling has been changed. Please redial on whatever the new number might be." It makes eminent sense - simple, common causes (e.g. user busy) get dealt with as part of the call clearing and it's the responsibility of the originating switch to tell the user; more (indeed arbitrarily) complex ones are dealt with by the far end. --Dave From carthick84 at gmail.com Mon Apr 6 07:04:11 2009 From: carthick84 at gmail.com (B Karthik) Date: Mon, 6 Apr 2009 19:34:11 +0530 Subject: [Freeswitch-users] High CPU load but only few sessions Message-ID: It could be due to registrations. I am currently trying to troubleshoot this problem. I used a sipp scenario to authenticate with fs and register about 2000 different accounts (absolutely no calls made on the test setup). Memory usage increases continuously and does not decrease at all and crosses more than 1 GB in a few hours. On the other hand, there is another fs setup with bypass media turned on and no registrations and is up for almost 45 days without restart and has consumed only about 95 MB of memory and twice as much virtual memory. B Karthik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/75a24d68/attachment.html From brian at freeswitch.org Mon Apr 6 07:14:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Apr 2009 09:14:15 -0500 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: References: Message-ID: <6703B4CB-4097-4349-9427-D5B19C6474E7@freeswitch.org> If you guys are not on rev 12914 then you'll need to update. /b On Apr 6, 2009, at 9:04 AM, B Karthik wrote: > It could be due to registrations. I am currently trying to > troubleshoot this problem. I used a sipp scenario to authenticate > with fs and register about 2000 different accounts (absolutely no > calls made on the test setup). Memory usage increases continuously > and does not decrease at all and crosses more than 1 GB in a few > hours. On the other hand, there is another fs setup with bypass > media turned on and no registrations and is up for almost 45 days > without restart and has consumed only about 95 MB of memory and > twice as much virtual memory. > > B Karthik Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/a2fa2196/attachment.html From helmut.kuper at ewetel.de Mon Apr 6 07:32:05 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 06 Apr 2009 16:32:05 +0200 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: References: Message-ID: <49DA1265.4050907@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, in my scenario I have a reregistration interval of 60 seconds and 32 sip phones connected. So I have a good amount of registrations. Additionally each phone subscribes to itself for MWI and some phone subscribes to others for BLF. Registrar database looks fine. No unused entries there. First I will upgrade to recent svn trunk. If that doesn't help, I will run valgrind on my production system and hope that my machine is strong enough to deliver its service even with valrgind. regards Helmut On 06.04.2009 16:04, B Karthik wrote: > It could be due to registrations. I am currently trying to troubleshoot > this problem. I used a sipp scenario to authenticate with fs and > register about 2000 different accounts (absolutely no calls made on the > test setup). Memory usage increases continuously and does not decrease > at all and crosses more than 1 GB in a few hours. On the other hand, > there is another fs setup with bypass media turned on and no > registrations and is up for almost 45 days without restart and has > consumed only about 95 MB of memory and twice as much virtual memory. > > B Karthik > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFJ2hJl4tZeNddg3dwRAmEUAKCTt1aBPlp1pgs3RHw2AVEuH8ixqgCfWkm8 6vsgh6Ha34/gdg6iDEEEOR0= =2H4m -----END PGP SIGNATURE----- From carthick84 at gmail.com Mon Apr 6 08:02:22 2009 From: carthick84 at gmail.com (B Karthik) Date: Mon, 6 Apr 2009 08:02:22 -0700 (PDT) Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49DA1265.4050907@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> Message-ID: <1239030142502-2593558.post@n2.nabble.com> I updated to the latest revision. No Luck -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, in my scenario I have a reregistration interval of 60 seconds and 32 sip phones connected. So I have a good amount of registrations. Additionally each phone subscribes to itself for MWI and some phone subscribes to others for BLF. Registrar database looks fine. No unused entries there. First I will upgrade to recent svn trunk. If that doesn't help, I will run valgrind on my production system and hope that my machine is strong enough to deliver its service even with valrgind. regards Helmut On 06.04.2009 16:04, B Karthik wrote: > It could be due to registrations. I am currently trying to troubleshoot > this problem. I used a sipp scenario to authenticate with fs and > register about 2000 different accounts (absolutely no calls made on the > test setup). Memory usage increases continuously and does not decrease > at all and crosses more than 1 GB in a few hours. On the other hand, > there is another fs setup with bypass media turned on and no > registrations and is up for almost 45 days without restart and has > consumed only about 95 MB of memory and twice as much virtual memory. > > B Karthik > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFJ2hJl4tZeNddg3dwRAmEUAKCTt1aBPlp1pgs3RHw2AVEuH8ixqgCfWkm8 6vsgh6Ha34/gdg6iDEEEOR0= =2H4m -----END PGP SIGNATURE----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593558.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Mon Apr 6 08:21:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 6 Apr 2009 10:21:36 -0500 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <1239030142502-2593558.post@n2.nabble.com> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> <1239030142502-2593558.post@n2.nabble.com> Message-ID: <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> did you both follow the policy to upgrade? stop fs type make current restart fs if you do not rebuild sofia too (only happens in make current) I just fixed all the problems with these symptoms, 38 million registrations in a 2 day span using 62mb btw, did we not make the policy clear enough about not reporting bugs on the mailing list? On Mon, Apr 6, 2009 at 10:02 AM, B Karthik wrote: > > I updated to the latest revision. No Luck > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > in my scenario I have a reregistration interval of 60 seconds and 32 sip > phones connected. So I have a good amount of registrations. Additionally > each phone subscribes to itself for MWI and some phone subscribes to > others for BLF. > > Registrar database looks fine. No unused entries there. > > First I will upgrade to recent svn trunk. If that doesn't help, I will > run valgrind on my production system and hope that my machine is strong > enough to deliver its service even with valrgind. > > regards > Helmut > > > On 06.04.2009 16:04, B Karthik wrote: > > It could be due to registrations. I am currently trying to troubleshoot > > this problem. I used a sipp scenario to authenticate with fs and > > register about 2000 different accounts (absolutely no calls made on the > > test setup). Memory usage increases continuously and does not decrease > > at all and crosses more than 1 GB in a few hours. On the other hand, > > there is another fs setup with bypass media turned on and no > > registrations and is up for almost 45 days without restart and has > > consumed only about 95 MB of memory and twice as much virtual memory. > > > > B Karthik > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFJ2hJl4tZeNddg3dwRAmEUAKCTt1aBPlp1pgs3RHw2AVEuH8ixqgCfWkm8 > 6vsgh6Ha34/gdg6iDEEEOR0= > =2H4m > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > View this message in context: > http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593558.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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/84fc6cba/attachment.html From carthick84 at gmail.com Mon Apr 6 08:30:41 2009 From: carthick84 at gmail.com (B Karthik) Date: Mon, 6 Apr 2009 08:30:41 -0700 (PDT) Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> <1239030142502-2593558.post@n2.nabble.com> <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> Message-ID: <1239031841283-2593704.post@n2.nabble.com> yes, i did exactly as you mentioned. I will try building again from a fresh checkout. I am sorry about not following the policy, I didn't intend to report it as a bug since i was still unsure that it could be a problem in Freeswitch. did you both follow the policy to upgrade? stop fs type make current restart fs if you do not rebuild sofia too (only happens in make current) I just fixed all the problems with these symptoms, 38 million registrations in a 2 day span using 62mb btw, did we not make the policy clear enough about not reporting bugs on the mailing list? On Mon, Apr 6, 2009 at 10:02 AM, B Karthik wrote: > > I updated to the latest revision. No Luck > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > in my scenario I have a reregistration interval of 60 seconds and 32 sip > phones connected. So I have a good amount of registrations. Additionally > each phone subscribes to itself for MWI and some phone subscribes to > others for BLF. > > Registrar database looks fine. No unused entries there. > > First I will upgrade to recent svn trunk. If that doesn't help, I will > run valgrind on my production system and hope that my machine is strong > enough to deliver its service even with valrgind. > > regards > Helmut > > > On 06.04.2009 16:04, B Karthik wrote: > > It could be due to registrations. I am currently trying to troubleshoot > > this problem. I used a sipp scenario to authenticate with fs and > > register about 2000 different accounts (absolutely no calls made on the > > test setup). Memory usage increases continuously and does not decrease > > at all and crosses more than 1 GB in a few hours. On the other hand, > > there is another fs setup with bypass media turned on and no > > registrations and is up for almost 45 days without restart and has > > consumed only about 95 MB of memory and twice as much virtual memory. > > > > B Karthik > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFJ2hJl4tZeNddg3dwRAmEUAKCTt1aBPlp1pgs3RHw2AVEuH8ixqgCfWkm8 > 6vsgh6Ha34/gdg6iDEEEOR0= > =2H4m > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > View this message in context: > http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593558.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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593704.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Mon Apr 6 08:36:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 6 Apr 2009 10:36:36 -0500 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <1239025636.12559.13.camel@dk-d820> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> <1239025636.12559.13.camel@dk-d820> Message-ID: <191c3a030904060836ka14d59ciaa89f1e7a768df8@mail.gmail.com> The default in originate is to return as soon as there is media. So if you bridge an inbound call, FS core will use originate to establish the outbound leg, as soon as it gets media (18X + sdp) it will return and enter the bridge in early media, this allows you to hear the early media while you are waiting for answer. If you want to wait for answer you add {ignore_early_media=true} to the dial string which tells originate to wait for answer or hangup before returning. if you are doing a forked dial and you don't just want the first one that has media to send a 183, you need to also enable {ignore_early_media=true} for that call. On Mon, Apr 6, 2009 at 8:47 AM, David Knell wrote: > On Mon, 2009-04-06 at 00:08 -0400, Kristian Kielhofner wrote: > > > Actually using 180 w/o SDP provides for enhanced call handing > > functionality while only requiring (in many cases) one additional test > > scenario. Consider the current example (all 180s are actually 180s > > w/o SDP and 183 is 183 w/ SDP): > > > > Bridging a call to multiple destinations (A, B, and C). > > > > A: 100,180 > > B: 100,180,200 > > C: 100,183 > > > > We could have implemented proper forking if it weren't for C who > > insisted on sending media early (for whatever reason). While I could > > see many scenarios where this might happen even with the configuration > > I suggest, consider what would happen in the ideal scenario: > > > > A: 100,180 > > B: 100,180,200 > > C: 100,180 > > > Ah, B won because it was the first endpoint to actually /answer/ the > > call and begin playing media. Nice and clean. > > Hang on - if you want to bridge the call on *answer*, then bridge it on > answer, not when one leg starts sending you early media. I've no idea > if FS supports this behaviour for its forked dialling, but it's easy > to do with a bunch of originates, and uuid_bridge the inbound leg to the > first one which answers. > > > People poke at SIP all the time for this one but this is where the > > PSTN even seems a bit ambiguous. We have ISDN cause codes AND inband > > audio messages? > > Yes. A clearing code is used when the call's cleared; inband audio > can be used to give the caller more information than a simple clearing > code might allow - for example, "The number you are calling has been > changed. Please redial on whatever the new number might be." It > makes eminent sense - simple, common causes (e.g. user busy) get dealt > with as part of the call clearing and it's the responsibility of the > originating switch to tell the user; more (indeed arbitrarily) complex > ones are dealt with by the far end. > > --Dave > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/6ff22ae1/attachment.html From codecomplete at free.fr Mon Apr 6 08:56:12 2009 From: codecomplete at free.fr (Fred) Date: Mon, 06 Apr 2009 17:56:12 +0200 Subject: [Freeswitch-users] Freeswitch not started OK at boottime? Message-ID: <7.0.1.0.2.20090406175519.024273d8@fredshack.com> Brian West-3 > What run level are you starting freeswitch? 3 to 5, the default being 5 (it's the desktop version, hence starting with X): # cat /etc/inittab [...] id:5:initdefault: # chkconfig -l freeswitch freeswitch 0:off 1:off 2:off 3:on 4:on 5:on 6:off dujinfang > It seems started, I never used a suse, however, can you try this? #netstat -an | grep 8021 # netstat -an | grep 8021 tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN > Maybe FS started before network is ready. Check scripts in /etc/rc.d/ or any equiv It looks ok: # ll /etc/rc.d/rc5.d/ [...] lrwxrwxrwx 1 root root 10 Jun 16 2008 S05network -> ../network [...] lrwxrwxrwx 1 root root 13 Mar 24 16:36 S12freeswitch -> ../freeswitch lrwxrwxrwx 1 root root 6 Jun 16 2008 S12xdm -> ../xdm lrwxrwxrwx 1 root root 8 Jun 16 2008 S14smbfs -> ../smbfs lrwxrwxrwx 1 root root 13 Jun 16 2008 S15cupsrenice -> ../cupsrenice If it's a Suse-specific issue, I'll go ask in a Suse forum and see if someone can figure it out. Thanks guys for the tips. From carthick84 at gmail.com Mon Apr 6 09:06:01 2009 From: carthick84 at gmail.com (B Karthik) Date: Mon, 6 Apr 2009 09:06:01 -0700 (PDT) Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <1239031841283-2593704.post@n2.nabble.com> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> <1239030142502-2593558.post@n2.nabble.com> <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> <1239031841283-2593704.post@n2.nabble.com> Message-ID: <1239033961503-2593907.post@n2.nabble.com> Great work. Memory usage is constant now. Memory is now Res :162M Virt: 483M for more than 10 mins without increasing. Call rate was set to 100 in sipp. However "top" usage is very high - 137% - 200%. MySQL usage is about 3% constant. I will also try overriding the XML bind function with a function which can lookup an in memory cached hash table for directory entries. I will also try by disabling mysql and post the results shortly. yes, i did exactly as you mentioned. I will try building again from a fresh checkout. I am sorry about not following the policy, I didn't intend to report it as a bug since i was still unsure that it could be a problem in Freeswitch. did you both follow the policy to upgrade? stop fs type make current restart fs if you do not rebuild sofia too (only happens in make current) I just fixed all the problems with these symptoms, 38 million registrations in a 2 day span using 62mb btw, did we not make the policy clear enough about not reporting bugs on the mailing list? On Mon, Apr 6, 2009 at 10:02 AM, B Karthik wrote: > > I updated to the latest revision. No Luck > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > in my scenario I have a reregistration interval of 60 seconds and 32 sip > phones connected. So I have a good amount of registrations. Additionally > each phone subscribes to itself for MWI and some phone subscribes to > others for BLF. > > Registrar database looks fine. No unused entries there. > > First I will upgrade to recent svn trunk. If that doesn't help, I will > run valgrind on my production system and hope that my machine is strong > enough to deliver its service even with valrgind. > > regards > Helmut > > > On 06.04.2009 16:04, B Karthik wrote: > > It could be due to registrations. I am currently trying to troubleshoot > > this problem. I used a sipp scenario to authenticate with fs and > > register about 2000 different accounts (absolutely no calls made on the > > test setup). Memory usage increases continuously and does not decrease > > at all and crosses more than 1 GB in a few hours. On the other hand, > > there is another fs setup with bypass media turned on and no > > registrations and is up for almost 45 days without restart and has > > consumed only about 95 MB of memory and twice as much virtual memory. > > > > B Karthik > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFJ2hJl4tZeNddg3dwRAmEUAKCTt1aBPlp1pgs3RHw2AVEuH8ixqgCfWkm8 > 6vsgh6Ha34/gdg6iDEEEOR0= > =2H4m > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > View this message in context: > http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593558.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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593907.html Sent from the freeswitch-users mailing list archive at Nabble.com. From helmut.kuper at ewetel.de Mon Apr 6 09:09:12 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 06 Apr 2009 18:09:12 +0200 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> <1239030142502-2593558.post@n2.nabble.com> <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> Message-ID: <49DA2928.9030205@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Anthony, I did a fresh checkout, compiled it, installed it into a clean directory and will switch over to it tomorrow morning. I hope I can reuse this existing directories: db/ conf/ storage/ sounds/ On 06.04.2009 17:21, Anthony Minessale wrote: > btw, > did we not make the policy clear enough about not reporting bugs on the > mailing list? Yes you did ... I apologize! regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFJ2iko4tZeNddg3dwRAnfWAKCHz7MeJZscWPLMkKQV6lflp8Wi+gCePRQm Cz3J66fWtZEMK+n7D8GXAM8= =mFi1 -----END PGP SIGNATURE----- From andy at fabulous4.co.uk Mon Apr 6 09:10:06 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 6 Apr 2009 17:10:06 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: Message-ID: <517C212F5DE6459181AE16036E04179F@wsandy> Hi Brian, Ok, all up to date, the errors have gone and the software is basically working but the cut off problem still exists. I have an identical software install running on a machine that is not behind a firewall and the cut off doesn't seem to occur. This would seem to suggest it's firewall related. Any clues? regards Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 14:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please update... rebootstrap.. you caught SVN with the libtool patch which kinda broken a few things linking. /b On Apr 6, 2009, at 4:07 AM, Andy Ayers wrote: Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/78947bac/attachment-0001.html From brian at freeswitch.org Mon Apr 6 09:25:46 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Apr 2009 11:25:46 -0500 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <517C212F5DE6459181AE16036E04179F@wsandy> References: <517C212F5DE6459181AE16036E04179F@wsandy> Message-ID: <70CD2E47-AB93-46E7-9198-ED2F5FEBC264@freeswitch.org> Don't record in Mp3, I don't recommend it.. /b On Apr 6, 2009, at 11:10 AM, Andy Ayers wrote: > Hi Brian, > > Ok, all up to date, the errors have gone and the software is > basically working but the cut off problem still exists. I have an > identical software install running on a machine that is not behind a > firewall and the cut off doesn't seem to occur. This would seem to > suggest it's firewall related. Any clues? > > regards > Andy > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/95669636/attachment.html From andy at fabulous4.co.uk Mon Apr 6 09:31:47 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 6 Apr 2009 17:31:47 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: Message-ID: <2D49B31443C64C1AB49E9264C71A5A21@wsandy> Hi Brian, Just doing some more testing, simplified the call by not even trying to record the incoming audio and placing a while (session.ready()) {} loop in the ivr code instead and the calls all now terminate with RECOVERY_ON_TIMER_EXPIRE. Does this shed any light on the subject at all? regards Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 14:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please update... rebootstrap.. you caught SVN with the libtool patch which kinda broken a few things linking. /b On Apr 6, 2009, at 4:07 AM, Andy Ayers wrote: Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/d6ddf6af/attachment.html From brian at freeswitch.org Mon Apr 6 09:39:11 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Apr 2009 11:39:11 -0500 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <2D49B31443C64C1AB49E9264C71A5A21@wsandy> References: <2D49B31443C64C1AB49E9264C71A5A21@wsandy> Message-ID: <581BF129-4E4B-4BBD-9F14-7EB88138101F@freeswitch.org> Three letters come to mind... N A T! ;) What is your network topo? /b On Apr 6, 2009, at 11:31 AM, Andy Ayers wrote: > Hi Brian, > > Just doing some more testing, simplified the call by not even trying > to record the incoming audio and placing a while (session.ready()) > {} loop in the ivr code instead and the calls all now terminate with > RECOVERY_ON_TIMER_EXPIRE. > > Does this shed any light on the subject at all? > > regards > Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/efe51686/attachment.html From mszlazak at aol.com Sun Apr 5 10:14:32 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sun, 05 Apr 2009 13:14:32 -0400 Subject: [Freeswitch-users] MANDATORY_IE_MISSING and NORMAL_TEMPORARY_FAILURE using javascript in dialplan. Message-ID: <8CB84291EE83AD4-D1C-FC0@webmail-db21.sysops.aol.com> On Windows XP/SP3 with FS trunk 12653M I get these errors using javascript in my dialplan: ? [MANDATORY_IE_MISSING] (see pastebin below) and/or with [CS_EXCHANGE_MEDIA] I get [NORMAL_TEMPORARY_FAILURE]? (not shown this time in pastebin) Here is the test javascript file: session.answer(); session.setVariable("choice", "demo"); If I remove session.answer() in the above test script then there is no problem but that doesn't always work with another scripts. Here is the related section of the dialplan. You dial into ext. 2222 and the problem happens at . The variable "choice" is set to "demo" to get here. ? ??? ??? ? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ? ??? ? ??? ??? ??? ??? ??? ??? ??? ? ??? ??? ? ??? ??? ??? ??? ??? ??? ?? ??? ??? ??? ? ??? ??? ? ??? ? ??? ? ??? ??? ? ??? ??? ?? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ?? ??? ??? ??? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ? ??? ? ??? ? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ? ??? I enabled SIP/Sofia tracing and "pastebinned" part of the output here: ?http://pastebin.freeswitch.org/8321 Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090405/f000a858/attachment-0001.html From andy at fabulous4.co.uk Mon Apr 6 09:52:51 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 6 Apr 2009 17:52:51 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <581BF129-4E4B-4BBD-9F14-7EB88138101F@freeswitch.org> Message-ID: Hi Brian, The freeswitch server is connect to the internet via a Cisico ASA firewall currently running in NAT mode. I believe it's that simple but can't be sure of the equipment between my firewall and the internet. regards Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 17:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Three letters come to mind... N A T! ;) What is your network topo? /b On Apr 6, 2009, at 11:31 AM, Andy Ayers wrote: Hi Brian, Just doing some more testing, simplified the call by not even trying to record the incoming audio and placing a while (session.ready()) {} loop in the ivr code instead and the calls all now terminate with RECOVERY_ON_TIMER_EXPIRE. Does this shed any light on the subject at all? regards Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/a106bb4b/attachment.html From mike at jerris.com Mon Apr 6 13:03:42 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 6 Apr 2009 16:03:42 -0400 Subject: [Freeswitch-users] libtool 2.2 patch Message-ID: <0CFA8FFA-4621-4133-A0A4-C18EA55BDC99@jerris.com> I need some testers for systems using both libtool 2.2 and 1.5.x to confirm the following patch: http://jira.freeswitch.org/secure/attachment/11356/fs-r12922-libtool22.patch In order to test you will need to do a complete fresh checkout, apply this patch, then do a bootstrap, configure, etc. Please make sure both mod_spidermonkey and mod_shout both build AND load (you will need to edit modules.conf and modules.conf.xml for these modules) when you start freeswitch without any errors. Please post your test results to http://jira.freeswitch.org/browse/FSBUILD-82 Thanks Mike From fax at virgintechnologies.com Mon Apr 6 14:54:05 2009 From: fax at virgintechnologies.com (Justin Miller) Date: Mon, 06 Apr 2009 21:54:05 +0000 Subject: [Freeswitch-users] Native G729 file playback and recording in Windows Message-ID: I know there is an implementation of this for linux. Does anyone have it working in Windows? I gave it a try, but had no luck. I can get individual G729 files to play through the dialplan, but I couldn't get voicemail to work. Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/81a9099d/attachment.html From mszlazak at aol.com Mon Apr 6 15:42:39 2009 From: mszlazak at aol.com (mszlazak) Date: Mon, 6 Apr 2009 15:42:39 -0700 (PDT) Subject: [Freeswitch-users] MANDATORY_IE_MISSING and NORMAL_TEMPORARY_FAILURE using javascript in dialplan. In-Reply-To: <8CB84291EE83AD4-D1C-FC0@webmail-db21.sysops.aol.com> References: <8CB84291EE83AD4-D1C-FC0@webmail-db21.sysops.aol.com> Message-ID: <22918984.post@talk.nabble.com> I svn'd to today's latest trunk to see if the problem remained. However, things seemed to have turned worse. The error messages I had before don't occur but I still can't bridge to my other application with choice=demo. I tried the application by dialing straight into it with the following dial plan. This has worked in FS 1.0.1 through 1.0.3 but fails to make any connection in the current svn trunk. NOTE ON UNRELATED ERROR: I don't use Lua but there was an error in compiling LUA on Windows with 2008 Express so I get a error loading mod_lua.dll today. Simple dialpan that worked before: SIP TRACE: http://pastebin.freeswitch.org/8336 -- View this message in context: http://www.nabble.com/MANDATORY_IE_MISSING-and-NORMAL_TEMPORARY_FAILURE-using-javascript-in-dialplan.-tp22912880p22918984.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Mon Apr 6 15:47:32 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 6 Apr 2009 18:47:32 -0400 Subject: [Freeswitch-users] MANDATORY_IE_MISSING and NORMAL_TEMPORARY_FAILURE using javascript in dialplan. In-Reply-To: <22918984.post@talk.nabble.com> References: <8CB84291EE83AD4-D1C-FC0@webmail-db21.sysops.aol.com> <22918984.post@talk.nabble.com> Message-ID: <8F7C08B5-FF31-4787-877C-E138DCF64E0C@jerris.com> On Apr 6, 2009, at 6:42 PM, mszlazak wrote: > > NOTE ON UNRELATED ERROR: I don't use Lua but there was an error in > compiling > LUA on Windows with 2008 Express so I get a error loading > mod_lua.dll today. This is fixed in svn a bit earlier today. Mike From trevor at concipient.net Mon Apr 6 16:11:31 2009 From: trevor at concipient.net (Trevor Hammonds) Date: Mon, 6 Apr 2009 16:11:31 -0700 Subject: [Freeswitch-users] libtool 2.2 patch In-Reply-To: <0CFA8FFA-4621-4133-A0A4-C18EA55BDC99@jerris.com> References: <0CFA8FFA-4621-4133-A0A4-C18EA55BDC99@jerris.com> Message-ID: <711825c70904061611m5c26c500o2b1cae3c86ed2cb8@mail.gmail.com> When I attempt to apply the patch to rev 12932, it says that the patch is already detected. Has this already been merged? Sincerely, Trevor Hammonds On Mon, Apr 6, 2009 at 1:03 PM, Michael Jerris wrote: > I need some testers for systems using both libtool 2.2 and 1.5.x to > confirm the following patch: > > > http://jira.freeswitch.org/secure/attachment/11356/fs-r12922-libtool22.patch > > In order to test you will need to do a complete fresh checkout, apply > this patch, then do a bootstrap, configure, etc. > > Please make sure both mod_spidermonkey and mod_shout both build AND > load (you will need to edit modules.conf and modules.conf.xml for > these modules) when you start freeswitch without any errors. Please > post your test results to http://jira.freeswitch.org/browse/FSBUILD-82 > > Thanks > Mike > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/3a7b9a7e/attachment.html From diego.viola at gmail.com Mon Apr 6 16:31:05 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 6 Apr 2009 19:31:05 -0400 Subject: [Freeswitch-users] libtool 2.2 patch In-Reply-To: <711825c70904061611m5c26c500o2b1cae3c86ed2cb8@mail.gmail.com> References: <0CFA8FFA-4621-4133-A0A4-C18EA55BDC99@jerris.com> <711825c70904061611m5c26c500o2b1cae3c86ed2cb8@mail.gmail.com> Message-ID: <86a32abc0904061631k530b3ce3j54d489c032f030d2@mail.gmail.com> Hi Trevor, The patch has been merged on latest trunk already. Regards, Diego V. On Mon, Apr 6, 2009 at 7:11 PM, Trevor Hammonds wrote: > When I attempt to apply the patch to rev 12932, it says that the patch is > already detected. Has this already been merged? > > Sincerely, > Trevor Hammonds > > > On Mon, Apr 6, 2009 at 1:03 PM, Michael Jerris wrote: > >> I need some testers for systems using both libtool 2.2 and 1.5.x to >> confirm the following patch: >> >> >> http://jira.freeswitch.org/secure/attachment/11356/fs-r12922-libtool22.patch >> >> In order to test you will need to do a complete fresh checkout, apply >> this patch, then do a bootstrap, configure, etc. >> >> Please make sure both mod_spidermonkey and mod_shout both build AND >> load (you will need to edit modules.conf and modules.conf.xml for >> these modules) when you start freeswitch without any errors. Please >> post your test results to http://jira.freeswitch.org/browse/FSBUILD-82 >> >> Thanks >> 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/20090406/48c0c601/attachment.html From trevor at concipient.net Mon Apr 6 16:42:29 2009 From: trevor at concipient.net (Trevor Hammonds) Date: Mon, 6 Apr 2009 16:42:29 -0700 Subject: [Freeswitch-users] libtool 2.2 patch In-Reply-To: <86a32abc0904061631k530b3ce3j54d489c032f030d2@mail.gmail.com> References: <0CFA8FFA-4621-4133-A0A4-C18EA55BDC99@jerris.com> <711825c70904061611m5c26c500o2b1cae3c86ed2cb8@mail.gmail.com> <86a32abc0904061631k530b3ce3j54d489c032f030d2@mail.gmail.com> Message-ID: <711825c70904061642h27f19572mdf640e95e45b1dda@mail.gmail.com> Thanks! On Mon, Apr 6, 2009 at 4:31 PM, Diego Viola wrote: > Hi Trevor, > > The patch has been merged on latest trunk already. > > Regards, > > Diego V. > > On Mon, Apr 6, 2009 at 7:11 PM, Trevor Hammonds wrote: > >> When I attempt to apply the patch to rev 12932, it says that the patch is >> already detected. Has this already been merged? >> >> Sincerely, >> Trevor Hammonds >> >> >> On Mon, Apr 6, 2009 at 1:03 PM, Michael Jerris wrote: >> >>> I need some testers for systems using both libtool 2.2 and 1.5.x to >>> confirm the following patch: >>> >>> >>> http://jira.freeswitch.org/secure/attachment/11356/fs-r12922-libtool22.patch >>> >>> In order to test you will need to do a complete fresh checkout, apply >>> this patch, then do a bootstrap, configure, etc. >>> >>> Please make sure both mod_spidermonkey and mod_shout both build AND >>> load (you will need to edit modules.conf and modules.conf.xml for >>> these modules) when you start freeswitch without any errors. Please >>> post your test results to http://jira.freeswitch.org/browse/FSBUILD-82 >>> >>> Thanks >>> Mike >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/43b79b99/attachment-0001.html From mattdfong at gmail.com Mon Apr 6 23:52:54 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 7 Apr 2009 13:52:54 +0700 Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? Message-ID: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> I'm doing some outbound dialing, and want to use mod_vmd to detect if a live person picks up or a voicemail picks up. I've read the wiki, and have been playing around with the dialplan implementation and the lua implementation, along with capturing the mod_vmdvmd::beep event. Using the examples on the wiki, I am able to call a number, sleep for 25 seconds, and mod_vmd usually detects a Beep (the answering machine beep right before you are to speak your message). My question is, is there a way to use mod_vmd to detect if an answering machine or human has picked up within the first 1-2 seconds after being answered? If so, can I get an example of how to set this up? my dialplan to test my lua implementation looks like and matt_vmd.lua looks like print ("--matt_vmd.lua START--") local human_detected = false; local voicemail_detected = false; function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '1' and human_detected == false then print('MATT--I detected a HUMAN'); human_detected = true; return "break"; end if type == "event" and voicemail_detected == false then print('MATT--I detected a VOICEMAIL'); voicemail_detected = true; return "break"; end end session:setInputCallback("onInput"); session:execute("vmd"); session:sleep(25000); print ("--matt_vmd.lua FINISHED--") Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/d8d30c16/attachment.html From kristian.kielhofner at gmail.com Tue Apr 7 00:10:40 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 7 Apr 2009 03:10:40 -0400 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: <87f2f3b90904030930r2b82a5c3oa9c558b4c5f7052e@mail.gmail.com> References: <87f2f3b90904030930r2b82a5c3oa9c558b4c5f7052e@mail.gmail.com> Message-ID: <2d9149cd0904070010g20eafa1i7bdb513c824c26e6@mail.gmail.com> 2009/4/3 Michael Collins : > On Fri, Apr 3, 2009 at 7:11 AM, Brian West wrote: >> >> Did it sound more like a machine gun? >> /b > > Comfort noise for General Douglas McArthur I guess... > I thought General Norman Scwarzkopf (Stormin' Norman) would have been more appropriate: http://en.wikipedia.org/wiki/Norman_Schwarzkopf,_Jr. Sorry, I just couldn't resist. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From krice at freeswitch.org Tue Apr 7 00:19:21 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 07 Apr 2009 02:19:21 -0500 Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? In-Reply-To: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> Message-ID: Matt, No that?s all mod_vmd does... If you want to do a more advanced analysis of media stream coming from the client mod_amd is available under a commercial license. This does media analysis to determine machine vs humans based on a hand full of metrics that are tunable. Contact me off list for licensing details Ken Rice krice at freeswitch.org From: Matthew Fong Reply-To: Date: Tue, 7 Apr 2009 13:52:54 +0700 To: Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? I'm doing some outbound dialing, and want to use mod_vmd to detect if a live person picks up or a voicemail picks up. I've read the wiki, and have been playing around with the dialplan implementation and the lua implementation, along with capturing the mod_vmd vmd::beep event. Using the examples on the wiki, I am able to call a number, sleep for 25 seconds, and mod_vmd usually detects a Beep (the answering machine beep right before you are to speak your message). My question is, is there a way to use mod_vmd to detect if an answering machine or human has picked up within the first 1-2 seconds after being answered? If so, can I get an example of how to set this up? my dialplan to test my lua implementation looks like and matt_vmd.lua looks like print ("--matt_vmd.lua START--") local human_detected = false; local voicemail_detected = false; function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '1' and human_detected == false then print('MATT--I detected a HUMAN'); human_detected = true; return "break"; end if type == "event" and voicemail_detected == false then print('MATT--I detected a VOICEMAIL'); voicemail_detected = true; return "break"; end end session:setInputCallback("onInput"); session:execute("vmd"); session:sleep(25000); print ("--matt_vmd.lua FINISHED--") 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/f00f0808/attachment.html From mbrancaleoni at voismart.it Tue Apr 7 01:13:09 2009 From: mbrancaleoni at voismart.it (Matteo) Date: Tue, 7 Apr 2009 10:13:09 +0200 (CEST) Subject: [Freeswitch-users] Skype interaction commands on skypiax In-Reply-To: <7b197bef0904020135j6b56662dy5a0dd2862ac4f35d@mail.gmail.com> Message-ID: <573982303.40161239091989637.JavaMail.root@mx.voismart.com> Ciao Giovanni, I suggest to update the startskype.sh script by adding a "su username" statement, in this way: instead of starting skype as echo "myskypeuser xxx" | DISPLAY=:101 /usr/bin/skype --pipelogin & is better to do: su unixusername -c "echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype --pipelogin &" for two reason: you can easily put config into a non-root user AND the startskype.sh will work also if called from init. in fact, a plain echo "myskypeuser xxx" | DISPLAY=:101 /usr/bin/skype --pipelogin & will not work when called from init script, you have to do (even with root) su root -c "echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype --pipelogin &" in any other way skype will not get the user home directory... This is my 2c experience on centos 5.2. regards, matteo. From andy at fabulous4.co.uk Tue Apr 7 06:10:19 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 7 Apr 2009 14:10:19 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <581BF129-4E4B-4BBD-9F14-7EB88138101F@freeswitch.org> Message-ID: <6F855374D4C44C089862BD60E220EB54@wsandy> Hi Brian, Is NAT a known problem? Is there a work around? The messages on the lists seem to imply other folks have this working ok behind NAT firewalls. What's your recommendation for how I should proceed? regards Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 17:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Three letters come to mind... N A T! ;) What is your network topo? /b On Apr 6, 2009, at 11:31 AM, Andy Ayers wrote: Hi Brian, Just doing some more testing, simplified the call by not even trying to record the incoming audio and placing a while (session.ready()) {} loop in the ivr code instead and the calls all now terminate with RECOVERY_ON_TIMER_EXPIRE. Does this shed any light on the subject at all? regards Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/1b22b3f8/attachment-0001.html From jason at jasonjgw.net Tue Apr 7 00:17:06 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 7 Apr 2009 17:17:06 +1000 Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? In-Reply-To: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> References: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> Message-ID: <20090407071706.GA529@jdc.jasonjgw.net> Matthew Fong wrote: > My question is, is there a way to use mod_vmd to detect if an answering > machine or human has picked up within the first 1-2 seconds after being > answered? Probably not. If you have an algorithm in mind that would achieve this with a high degree of reliability, I'm sure the FreeSWITCH developers would be interested in it. However, as far as I know, there is no reliable way to distinguish, for example, my voice as recorded in a voicemail greeting from my voice giving a live greeting after answering a phone call. Think about it. From freeswitch at philstyle.com Mon Apr 6 19:51:39 2009 From: freeswitch at philstyle.com (Drew Ozier) Date: Mon, 6 Apr 2009 22:51:39 -0400 Subject: [Freeswitch-users] Problem listening to PCMU and ul recordings Message-ID: <2388e50e0904061951p1410dc6ayb2b5994009aaaa20@mail.gmail.com> Hi, I've been using the record_session feature and wish to use PCMU or ul audio format, but when I try to play back the audio in either format, it sounds high-pitch and fast as if it is playing back at 2x speed. I looked at the waveform recorded in PCMU and ul versus what it looks like when I record as wav, and it seems like it is only recording every-other sample (which would explain the pitch and speed). My vars.xml is set to PCMU as the global codec pref and the outbound codec pref. I am recording in stereo (one channel per leg of the call), but I'm not messing with any other recording parameters. Incidentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio file, because I'm getting calls off a T1 (actually off of an AudioCodes Mediant 1000 that is converting the T1 to SIP for me), and I'd like to record precisely what is coming off the wire. I'd be happy to send any configuration files, I'm just currently at a loss for how to proceed. Thanks, Drew Ozier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/fbf8341b/attachment.html From dschwartz at xconnect.net Tue Apr 7 06:45:53 2009 From: dschwartz at xconnect.net (David Schwartz) Date: Tue, 7 Apr 2009 14:45:53 +0100 Subject: [Freeswitch-users] Can anyone recommend any provisioning tools for use with FS? Message-ID: <062B8EE81F2EC945A577C3EFAE1DD68E99632F@mail.xconnect.net> Preferably GUI based. Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/658f95b4/attachment.html From pawzlion at gmail.com Tue Apr 7 06:46:57 2009 From: pawzlion at gmail.com (David Robinson) Date: Tue, 07 Apr 2009 23:46:57 +1000 Subject: [Freeswitch-users] problems with Faktortel (AU) and multiple DID's and extensions In-Reply-To: References: Message-ID: <49DB5951.2090006@gmail.com> I have been trying to setup 2 DID's to route to 2 extensions but whenever I try it, the second configured DID always routes to the first extension. In my public.xml I have the following: .... rest of file continues here ... While in my default.xml I have this: .. file continues here ... I got my new friend swk to try and diagnose the problem and using ngrep he found with ngrep that the incoming call to the second extension looked like this: U 203.161.130.133:5060 -> 10.0.0.12:5080 INVITE sip:gw+voicepulse at 10.0.0.12:5080;transport=udp SIP/2.0..Via: SIP/2.0/UDP 203.161.130.133:5060;branch=z9hG4bK75f53071;rport..From: "0451282630" ;tag=as555c5b50..To: ..Contact: ..Call-ID: 7 47befb63a2def723e6796294853cc22 at 203.161.130.133..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 07 Apr 2009 08:18:23 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Content-Type: application/sdp..Content-Length: 290....v=0..o=root 1244 12 44 IN IP4 203.161.130.133..s=session..c=IN IP4 203.161.130.133..t=0 0..m=audio 13806 RTP/AVP 18 3 101..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap :3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv.. He says that the INVITE line should have a DNIS (not sure what that is) in that field to indicate which number to route it to but that for some reason, my provider (Faktortel in Australia) is not supplying that information. Does anyone know whether the problem is really at my provider's end or at my end, and if it's at my end, where ? thanks, pawz From dave at 3c.co.uk Tue Apr 7 07:25:17 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 07 Apr 2009 15:25:17 +0100 Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? In-Reply-To: <20090407071706.GA529@jdc.jasonjgw.net> References: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> <20090407071706.GA529@jdc.jasonjgw.net> Message-ID: <1239114317.16460.16.camel@dk-d820> On Tue, 2009-04-07 at 17:17 +1000, Jason White wrote: > Matthew Fong wrote: > > My question is, is there a way to use mod_vmd to detect if an answering > > machine or human has picked up within the first 1-2 seconds after being > > answered? > > Probably not. If you have an algorithm in mind that would achieve this with a > high degree of reliability, I'm sure the FreeSWITCH developers would be > interested in it. However, as far as I know, there is no reliable way to > distinguish, for example, my voice as recorded in a voicemail greeting from my > voice giving a live greeting after answering a phone call. Think about it. The usual way is to measure how long the person who answers the phone speaks for. A person might say "Hello?", "Hello, this is Alice", "Thank you for calling XYZ. How may I direct your call?" Voicemail will usually be longer - "Hi, this is Bob. I'm sorry I can't take your call right now, so please leave me a message after the tone and I'll get back to you as soon as I can." In the first couple of cases above, this would give you an answer - "human" - within the first few seconds of the call. FreeSWITCH will give you TALK (start of speech (or noise)) and NOTALK (end of) events if you enable VAD. --Dave From brian at freeswitch.org Tue Apr 7 07:34:47 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Apr 2009 09:34:47 -0500 Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? In-Reply-To: <1239114317.16460.16.camel@dk-d820> References: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> <20090407071706.GA529@jdc.jasonjgw.net> <1239114317.16460.16.camel@dk-d820> Message-ID: <15A5B42A-454C-49FF-8895-B8DC2A01019F@freeswitch.org> Or my personal favorite... Congestion tone! /b On Apr 7, 2009, at 9:25 AM, David Knell wrote: > The usual way is to measure how long the person who answers the phone > speaks for. A person might say "Hello?", "Hello, this is Alice", > "Thank you for calling XYZ. How may I direct your call?" Voicemail > will usually be longer - "Hi, this is Bob. I'm sorry I can't take > your call right now, so please leave me a message after the tone and > I'll get back to you as soon as I can." Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/de00912a/attachment.html From freeswitch at philstyle.com Tue Apr 7 07:36:01 2009 From: freeswitch at philstyle.com (Drew Ozier) Date: Tue, 7 Apr 2009 10:36:01 -0400 Subject: [Freeswitch-users] Problem listening to PCMU and ul recordings Message-ID: <2388e50e0904070736i38213405q17e2b18fcf284779@mail.gmail.com> Hi, I've been using the record_session feature and wish to use PCMU or ul audio format, but when I try to play back the audio in either format, it sounds high-pitch and fast as if it is playing back at 2x speed. I looked at the waveform recorded in PCMU and ul versus what it looks like when I record as wav, and it seems like it is only recording every-other sample (which would explain the pitch and speed). My vars.xml is set to PCMU as the global codec pref and the outbound codec pref. I am recording in stereo (one channel per leg of the call), but I'm not messing with any other recording parameters. Incidentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio file, because I'm getting calls off a T1 (actually off of an AudioCodes Mediant 1000 that is converting the T1 to SIP for me), and I'd like to record precisely what is coming off the wire. I'd be happy to send any configuration files, I'm just currently at a loss for how to proceed. Thanks, Drew Ozier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/4cb19225/attachment.html From drew.ozier at gmail.com Tue Apr 7 07:51:20 2009 From: drew.ozier at gmail.com (Drew Ozier) Date: Tue, 7 Apr 2009 10:51:20 -0400 Subject: [Freeswitch-users] Problem listening to PCMU and ul recordings Message-ID: <2388e50e0904070751v606ae262uee0092db66dbcff5@mail.gmail.com> Hi, I've been using the record_session feature and wish to use PCMU or ul audio format, but when I try to play back the audio in either format, it sounds high-pitch and fast as if it is playing back at 2x speed. I looked at the waveform recorded in PCMU and ul versus what it looks like when I record as wav, and it seems like it is only recording every-other sample (which would explain the pitch and speed). My vars.xml is set to PCMU as the global codec pref and the outbound codec pref. I am recording in stereo (one channel per leg of the call), but I'm not messing with any other recording parameters. Incedentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio file, because I'm getting calls off a T1 (actually off of an AudioCodes Mediant 1000 that is converting the T1 to SIP for me), and I'd like to record precicely what is coming off the wire. I'd be happy to send any configuration files, I'm just currently at a loss for how to proceed. Thanks, Drew Ozier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/2ee503a7/attachment-0001.html From gmaruzz at celliax.org Tue Apr 7 09:32:16 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 7 Apr 2009 18:32:16 +0200 Subject: [Freeswitch-users] Skype interaction commands on skypiax In-Reply-To: <573982303.40161239091989637.JavaMail.root@mx.voismart.com> References: <7b197bef0904020135j6b56662dy5a0dd2862ac4f35d@mail.gmail.com> <573982303.40161239091989637.JavaMail.root@mx.voismart.com> Message-ID: <7b197bef0904070932p6464b6f5k226dabbdd0cb6c66@mail.gmail.com> svn commit -m"skypiax: modified configs/startskype.sh to specify which unix user will start the Skype client instance. Thx to mbrancaleoni at voismart.it" Sending configs/startskype.sh Transmitting file data . Committed revision 12937. :-) On Tue, Apr 7, 2009 at 10:13 AM, Matteo wrote: > Ciao Giovanni, > > I suggest to update the startskype.sh script by adding a "su username" statement, > in this way: > > instead of starting skype as > > echo "myskypeuser xxx" | DISPLAY=:101 /usr/bin/skype --pipelogin & > > is better to do: > > su unixusername -c "echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype --pipelogin &" > > for two reason: > you can easily put config into a non-root user > AND > the startskype.sh will work also if called from init. > > in fact, a plain > > echo "myskypeuser xxx" | DISPLAY=:101 /usr/bin/skype --pipelogin & > > will not work when called from init script, > you have to do (even with root) > > su root -c "echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype --pipelogin &" > > in any other way skype will not get the user home directory... > > This is my 2c experience on centos 5.2. > > regards, > matteo. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 7 10:19:16 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 07 Apr 2009 19:19:16 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption Message-ID: <49DB8B14.70400@gmx.net> I want to do the following: Due to missing Softclient with TLS/SRTP support on my Linux laptop (Zoiper is almost there, but not yet with SRTP) I want to install a local FS to listen on a local IP and then communicate via TLS/SRTP to my FS in the Office. As My Laptop has changing IPs (e.g. Ethernet-Cable, WLAN, UMTS) I want FS to listen on 127.0.0.1, connect my local VoIP client (Twinkle) to the local FS and communicate to the FS in my office through the public or LAN IP (maybe via STUN, public IP may change due to change of network connection). 1st Question: Is that possible or is another solution preferrable? 2nd Question: How can I change the amount of memory FS tries to reserve to an absolute minumum (I only have 1 call at a time). Currently it tries to reserve about 360M if I read that right. Best regards Peter From brian at freeswitch.org Tue Apr 7 10:33:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Apr 2009 12:33:06 -0500 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DB8B14.70400@gmx.net> References: <49DB8B14.70400@gmx.net> Message-ID: <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > 1st Question: Is that possible or is another solution preferrable? Just use FreeSWITCH with mod_portaudio. > 2nd Question: How can I change the amount of memory FS tries to > reserve > to an absolute minumum (I only have 1 call at a time). Currently it > tries to reserve about 360M if I read that right. Thats virtual. Look at RES. > > Best regards > Peter Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/91b69506/attachment.html From nik.middleton at noblesolutions.co.uk Tue Apr 7 15:32:13 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 7 Apr 2009 23:32:13 +0100 Subject: [Freeswitch-users] Hi Load, but calls still perfect Message-ID: Hi Guys, I'm no Linux guru, but today I inadvertently had 1000+ call attempts going through FS, load according to TOP was 16.5. Calls were still absolutely perfect. Can I throw out the rule book on load ? CPU was ~45% on each core. (dual) Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/92b49b11/attachment.html From Prometheus001 at gmx.net Tue Apr 7 15:52:54 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 00:52:54 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> References: <49DB8B14.70400@gmx.net> <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> Message-ID: <49DBD946.5060406@gmx.net> Thanks Brian, what I was actually looking for was to use a standard SIP soft phone with some additional features. I finally manged to make FS listen on 127.0.0.1 the following way: vars.xml internal.xml The rest is standard configuration. Now communication Laptop-internal is UDP on port 5060 and external via TLS on port 5081, so I have no open port 5060 to the internet. Best regards Peter Brian West schrieb: > > On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> 1st Question: Is that possible or is another solution preferrable? > > Just use FreeSWITCH with mod_portaudio. > >> 2nd Question: How can I change the amount of memory FS tries to reserve >> to an absolute minumum (I only have 1 call at a time). Currently it >> tries to reserve about 360M if I read that right. > > Thats virtual. Look at RES. > >> >> Best regards >> Peter > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 7 16:03:13 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 01:03:13 +0200 Subject: [Freeswitch-users] Conference fails with speex codec Message-ID: <49DBDBB1.3090404@gmx.net> I want to use a low bandwidth codec. But whenever I try to use speex I get an error in the conference. We have FS trunk 1288. Switching back to PCMx it works again. Is there any problem with speex and DTMF or with transcoding? Best regards Peter 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2624 sofia_glue_negotiate_sdp() Set Remote Key [1 AES_CM_128_HMAC_SHA1_32 inline:HU0NdX8n18lnRuEKhmJ1O4zSBaolz3wDtOwWIjy8] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:1917 sofia_glue_build_crypto() Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline:r98Rf+0lojftJOPPW9GZon5SZgB6Kg7FsED4cQV3] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[G722:9:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[PCMU:0:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[PCMA:8:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[GSM:3:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[SPEEX:98:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2786 sofia_glue_negotiate_sdp() Substituting codec SPEEX at 20i@16000h 2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error Opening File [/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav] [System error : No such file or directory.] 2009-04-08 00:43:24 [WARNING] mod_conference.c:4799 conference_function() Cannot ask the user for a pin, ending call2009-04-08 00:43:24 [NOTICE] mod_conference.c:4800 conference_function() Hangup sofia/internal/723328 at sip.mydomain.com [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] From msc at freeswitch.org Tue Apr 7 16:35:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Apr 2009 16:35:58 -0700 Subject: [Freeswitch-users] Conference fails with speex codec In-Reply-To: <49DBDBB1.3090404@gmx.net> References: <49DBDBB1.3090404@gmx.net> Message-ID: <87f2f3b90904071635y11d0ccd7gbfab0537cdfbb3d4@mail.gmail.com> > > > 2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error > Opening File > [/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav] > [System error : No such file or directory.] > 2009-04-08 00:43:24 [WARNING] mod_conference.c:4799 > conference_function() Cannot ask the user for a pin, ending This is curious. Do you see this error about the missing file when you use PCMU? -MC > > call2009-04-08 00:43:24 [NOTICE] mod_conference.c:4800 > conference_function() Hangup sofia/internal/723328 at sip.mydomain.com > [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/dad6c3bf/attachment.html From brian at freeswitch.org Tue Apr 7 16:52:12 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Apr 2009 18:52:12 -0500 Subject: [Freeswitch-users] Conference fails with speex codec In-Reply-To: <87f2f3b90904071635y11d0ccd7gbfab0537cdfbb3d4@mail.gmail.com> References: <49DBDBB1.3090404@gmx.net> <87f2f3b90904071635y11d0ccd7gbfab0537cdfbb3d4@mail.gmail.com> Message-ID: <3C6448A5-1D6A-4612-80A2-644CF0BE8F88@freeswitch.org> Chances are he just doesn't have the 16k sound files installed. /b On Apr 7, 2009, at 6:35 PM, Michael Collins wrote: > > 2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error > Opening File > [/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav] > [System error : No such file or directory.] > 2009-04-08 00:43:24 [WARNING] mod_conference.c:4799 > conference_function() Cannot ask the user for a pin, ending > > This is curious. Do you see this error about the missing file when > you use PCMU? > -MC > > > call2009-04-08 00:43:24 [NOTICE] mod_conference.c:4800 > conference_function() Hangup sofia/internal/723328 at sip.mydomain.com > [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/b36df24f/attachment-0001.html From jim at evolutiontel.net Tue Apr 7 17:13:13 2009 From: jim at evolutiontel.net (jim at evolutiontel.net) Date: Wed, 8 Apr 2009 10:13:13 +1000 Subject: [Freeswitch-users] problems with Faktortel (AU) and multiple D ID's and extensions Message-ID: <0cXkljJNrdVq.Y1qLYL1r@smtp.gmail.com> Hi David, Have seen a similar issue reported on whirlpool recently with another provider, essentially if the ITSP does not forward the To: header with the correct terminating DID you will not be able to determine the extention to route the call to. Am I correct in saying you only have one Faktortel account with 2 DIDs attached? Regards, Jim - original message - Subject: [Freeswitch-users] problems with Faktortel (AU) and multiple DID's and extensions From: David Robinson Date: 07/04/2009 13:50 I have been trying to setup 2 DID's to route to 2 extensions but whenever I try it, the second configured DID always routes to the first extension. In my public.xml I have the following: .... rest of file continues here ... While in my default.xml I have this: .. file continues here ... I got my new friend swk to try and diagnose the problem and using ngrep he found with ngrep that the incoming call to the second extension looked like this: U 203.161.130.133:5060 -> 10.0.0.12:5080 INVITE sip:gw+voicepulse at 10.0.0.12:5080;transport=udp SIP/2.0..Via: SIP/2.0/UDP 203.161.130.133:5060;branch=z9hG4bK75f53071;rport..From: "0451282630" ;tag=as555c5b50..To: ..Contact: ..Call-ID: 7 47befb63a2def723e6796294853cc22 at 203.161.130.133..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 07 Apr 2009 08:18:23 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Content-Type: application/sdp..Content-Length: 290....v=0..o=root 1244 12 44 IN IP4 203.161.130.133..s=session..c=IN IP4 203.161.130.133..t=0 0..m=audio 13806 RTP/AVP 18 3 101..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap :3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv.. He says that the INVITE line should have a DNIS (not sure what that is) in that field to indicate which number to route it to but that for some reason, my provider (Faktortel in Australia) is not supplying that information. Does anyone know whether the problem is really at my provider's end or at my end, and if it's at my end, where ? thanks, pawz _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 wiltingtree at gmail.com Tue Apr 7 17:35:56 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Tue, 7 Apr 2009 20:35:56 -0400 Subject: [Freeswitch-users] Two or more simultaneous calls not working Message-ID: Hello, I wrote an application using FreeSWITCH version 1.0.3, with mod_python and a 64 bit box on Red Hat. The app works fine when one person dials in, but when a second person dials in, the first call stops and waits until the second call is finished. It's really strange - if the first call is right in the middle of playing a prompt, it will just stop, and there will be dead air. As soon as the 2nd call hangs up, the prompt for the first call starts playing right where it left off. I previously had FreeSWITCH installed on a 32 bit CentOS box, and this was not happening. Does anybody have any idea what the cause of this could be? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/1c647dbd/attachment.html From john at feith.com Tue Apr 7 17:37:20 2009 From: john at feith.com (John Wehle) Date: Tue, 7 Apr 2009 20:37:20 -0400 (EDT) Subject: [Freeswitch-users] need help getting ISDN talking to Cisco 3845 Message-ID: <200904080037.n380bKih004889@jwlab.FEITH.COM> Our FreeSWITCH setup has an existing T1 using RBS to talk to a digital modem pack in a Cisco 3845. I'm interested in changing from RBS to ISDN. I changed both sides, restart things, and see FreeSWITCH report: 2009-04-07 18:53:15 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:54:40 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:55:36 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:55:45 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:46 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:47 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:48 [NOTICE] Span:0 Q.921() I frame in invalid state ignored I've attached the configs and Cisco debug below. This is using the native ISDN support in FreeSWITCH with a Sangoma A104d on FreeBSD 6.4. I unfortunately don't currently speak ISDN (though I'm starting to pick up a little as a result of this exercise) ... suggestions / hints regarding what's going on and how to resolve it would be welcomed. -- John ------------------------------ wanpipe2.conf ------------------------------- [devices] wanpipe2 = WAN_AFT_TE1, Comment [interfaces] wbg1 = wanpipe2, , TDM_VOICE, Comment [wanpipe2] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 5 PCIBUS = 5 FE_MEDIA = T1 FE_LCODE = B8ZS FE_FRAME = ESF FE_LINE = 2 TE_CLOCK = MASTER TE_REF_CLOCK = 1 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 2 TDMV_DCHAN = 0 TDMV_HW_DTMF = YES [wbg1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES ------------------------------ zaptel.conf --------------------------------- #Sangoma A104 port 2 [slot:5 bus:5 span:2] span=2,0,0,esf,b8zs bchan=25-47 dchan=48 ------------------------------ openzap.conf -------------------------------- [span zt] ; A104D FE 2 1-6 MICA name => Cisco Digital Modem trunk_type => t1 number => 2487 b-channel => 25-47 d-channel => 48 --------------------------- openzap.conf.xml ------------------------------- ------------------------------ Cisco config -------------------------------- controller T1 1/0 framing ESF linecode b8zs cablelength short 220 pri-group timeslots 1-24 interface Serial1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice modem isdn calling-number 2487 no cdp enable ------------------------------ Cisco debug --------------------------------- #show isdn stat Global ISDN Switchtype = primary-ni ISDN Serial1/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x807FFFFF Number of L2 Discards = 2, L2 Session ID = 117 Total Allocated ISDN CCBs = 0 Apr 7 22:53:44.264: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:53:44.264: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:53:44.312: ISDN Se1/0:23 Q921: User RX <- UAf sapi=0 tei=0 Apr 7 22:53:44.312: %CSM-5-PRI: add PRI at 1/0:23 (index 0) Apr 7 22:53:44.312: %ISDN-6-LAYER2UP: Layer 2 for Interface Se1/0:23, TEI 0 cha nged to up Apr 7 22:53:47.268: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:53:47.268: ISDN Se1/0:23 Q921: User TX -> RRf sapi=0 tei=0 nr=0 prepnet-rt# Apr 7 22:53:57.336: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:53:57.336: ISDN Se1/0:23 Q921: User TX -> RRf sapi=0 tei=0 nr=0 prepnet-rt# Apr 7 22:54:11.692: ISDN Se1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0 Apr 7 22:54:11.692: ISDN Se1/0:23 Q921: User TX -> UAf sapi=0 tei=0 Apr 7 22:54:21.760: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:54:21.760: ISDN Se1/0:23 Q921: User TX -> RRf sapi=0 tei=0 nr=0 Apr 7 22:54:51.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:52.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:53.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:54.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:55.760: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:54:55.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:56.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:57.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:58.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:59.760: %CSM-5-PRI: delete PRI at 1/0:23 (index 0) Apr 7 22:54:59.760: %ISDN-6-LAYER2DOWN: Layer 2 for Interface Se1/0:23, TEI 0 c hanged to down Apr 7 22:54:59.760: ISDN Se1/0:23 Q931: Ux_DLRelInd: DL_REL_IND received from L 2 Apr 7 22:55:04.760: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:55:04.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:05.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:05.872: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:55:06.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:07.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:08.760: ISDN Se1/0:23 Q931: Ux_DLRelInd: DL_REL_IND received from L 2 Apr 7 22:55:13.760: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:55:13.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:14.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:15.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:15.772: ISDN Se1/0:23 Q921: User RX <- UAf sapi=0 tei=0 Apr 7 22:55:15.772: %CSM-5-PRI: add PRI at 1/0:23 (index 0) Apr 7 22:55:15.772: %ISDN-6-LAYER2UP: Layer 2 for Interface Se1/0:23, TEI 0 cha nged to up Apr 7 22:55:15.772: ISDN Se1/0:23 Q921: User TX -> INFO sapi=0 tei=0, ns=0 nr=0 Apr 7 22:55:15.772: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:16.772: ISDN Se1/0:23 Q921: S7_T200_EXPIRY: VA = 0, VS = 1 Apr 7 22:55:16.772: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=0 nr= 0 Apr 7 22:55:16.772: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:17.772: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=0 nr= 0 Apr 7 22:55:17.772: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:17.784: ISDN Se1/0:23 Q921: User RX <- RRf sapi=0 tei=0 nr=1 Apr 7 22:55:45.773: ISDN Se1/0:23 Q921: User TX -> INFO sapi=0 tei=0, ns=1 nr=0 Apr 7 22:55:45.773: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:46.773: ISDN Se1/0:23 Q921: S7_T200_EXPIRY: VA = 1, VS = 2 Apr 7 22:55:46.773: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=1 nr= 0 Apr 7 22:55:46.773: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:47.773: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=1 nr= 0 Apr 7 22:55:47.773: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From mszlazak at aol.com Tue Apr 7 18:40:28 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 07 Apr 2009 21:40:28 -0400 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DBD946.5060406@gmx.net> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net> Message-ID: <8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a new trunk and I have to go through vars.xml, etc changing $${local_ip_v4} like you did. Is there a way to change $${local_ip_v4} in one place. That way one wouldn't have remember all the locations that it needs to be changed? -----Original Message----- From: Peter P GMX To: freeswitch-users at lists.freeswitch.org Sent: Tue, 7 Apr 2009 3:52 pm Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption Thanks Brian, what I was actually looking for was to use a standard SIP soft phone with some additional features. I finally manged to make FS listen on 127.0.0.1 the following way: vars.xml internal.xml The rest is standard configuration. Now communication Laptop-internal is UDP on port 5060 and external via TLS on port 5081, so I have no open port 5060 to the internet. Best regards Peter Brian West schrieb: > > On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> 1st Question: Is that possible or is another solution preferrable? > > Just use FreeSWITCH with mod_portaudio. > >> 2nd Question: How can I change the amount of memory FS tries to reserve >> to an absolute minumum (I only have 1 call at a time). Currently it >> tries to reserve about 360M if I read that right. > > Thats virtual. Look at RES. > >> >> Best regards >> Peter > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/7911bc28/attachment-0001.html From jason at jasonjgw.net Tue Apr 7 19:04:56 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 8 Apr 2009 12:04:56 +1000 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> References: <49DBD946.5060406@gmx.net> <8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> Message-ID: <20090408020456.GA11455@jdc.jasonjgw.net> mszlazak at aol.com wrote: > Is there a way to change $${local_ip_v4} in one place. Of course. That's why it's a variable. this goes in vars.xml, substituting the desired address. From tleyden at branchcut.com Tue Apr 7 19:06:27 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Wed, 8 Apr 2009 06:36:27 +0430 Subject: [Freeswitch-users] Two or more simultaneous calls not working Message-ID: > > > Hello, > > I wrote an application using FreeSWITCH version 1.0.3, with mod_python and > a > 64 bit box on Red Hat. > The app works fine when one person dials in, but when a second person dials > in, the first call stops and waits until the second call is finished. It's > really strange - if the first call is right in the middle of playing a > prompt, it will just stop, and there will be dead air. As soon as the 2nd > call hangs up, the prompt for the first call starts playing right where it > left off. > > I previously had FreeSWITCH installed on a 32 bit CentOS box, and this was > not happening. > > Does anybody have any idea what the cause of this could be? > > Thanks! I'm running mod_python fine with 64-bit (Ubuntu 8) and fs svn 12793, and have not seen that problem. Which version of python? Did you build it yourself or was it from a package? If its from a package, please provide the version of Red Hat you are using. One possible cause is that python was not compiled with multi-threading support. I don't know how to check that however .. googled around and didn't find anything. If you fire up your python interpreter and type "import threading" do you get an error? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/bd34aa75/attachment.html From mszlazak at aol.com Tue Apr 7 19:23:41 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 07 Apr 2009 22:23:41 -0400 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <20090408020456.GA11455@jdc.jasonjgw.net> References: <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <20090408020456.GA11455@jdc.jasonjgw.net> Message-ID: <8CB86082B2BA7F3-14E8-2609@FWM-D28.sysops.aol.com> Wonderful! Thank you sir. -----Original Message----- From: Jason White To: freeswitch-users at lists.freeswitch.org Sent: Tue, 7 Apr 2009 7:04 pm Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption mszlazak at aol.com wrote: > Is there a way to change $${local_ip_v4} in one place. Of course. That's why it's a variable. this goes in vars.xml, substituting the desired address. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/564fc12f/attachment.html From solko at gcdf.pl Tue Apr 7 22:54:02 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 07:54:02 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net> <8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> Message-ID: <49DC3BFA.4060200@gcdf.pl> mszlazak at aol.com pisze: > I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > new trunk and I have to go through vars.xml, etc changing > $${local_ip_v4} like you did. > > Is there a way to change $${local_ip_v4} in one place. That way one > wouldn't have remember all the locations that it needs to be changed? > My configuration is not updated when I compile new version and install it. Do you run FS with configuration path pointed to svn work dir? > -----Original Message----- > From: Peter P GMX > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 7 Apr 2009 3:52 pm > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > Thanks Brian, > > > > what I was actually looking for was to use a standard SIP soft phone > > with some additional features. > > > > I finally manged to make FS listen on 127.0.0.1 the following way: > > > > vars.xml > > > > > > internal.xml > > > > > > > > The rest is standard configuration. > > > > Now communication Laptop-internal is UDP on port 5060 and external via > > TLS on port 5081, so I have no open port 5060 to the internet. > > > > Best regards > > Peter > > > > > > > > > > Brian West schrieb: > >> > >> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> 1st Question: Is that possible or is another solution preferrable? > >> > >> Just use FreeSWITCH with mod_portaudio. > >> > >>> 2nd Question: How can I change the amount of memory FS tries to reserve > >>> to an absolute minumum (I only have 1 call at a time). Currently it > >>> tries to reserve about 360M if I read that right. > >> > >> Thats virtual. Look at RES. > >> > >>> > >>> Best regards > >>> Peter > >> > >> Brian West > >> brian at freeswitch.org > > >> > >> -- Meet us a ClueCon! http://www.cluecon.com > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > * > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 mszlazak at aol.com Tue Apr 7 23:43:30 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 08 Apr 2009 02:43:30 -0400 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DC3BFA.4060200@gcdf.pl> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl> Message-ID: <8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> I'm not quite sure what your asking. Are you saying that I could run the latest FS svn but in a way that uses my "older" configuration files? If so then I don't, and don't know how ... blush blush. If that's the easiest thing to do then please tell me how. Thanks. Mark. -----Original Message----- From: Szymon Olko To: freeswitch-users at lists.freeswitch.org Sent: Tue, 7 Apr 2009 10:54 pm Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption mszlazak at aol.com pisze: > I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > new trunk and I have to go through vars.xml, etc changing > $${local_ip_v4} like you did. > > Is there a way to change $${local_ip_v4} in one place. That way one > wouldn't have remember all the locations that it needs to be changed? > My configuration is not updated when I compile new version and install it. Do you run FS with configuration path pointed to svn work dir? > -----Original Message----- > From: Peter P GMX > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 7 Apr 2009 3:52 pm > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > Thanks Brian, > > > > what I was actually looking for was to use a standard SIP soft phone > > with some additional features. > > > > I finally manged to make FS listen on 127.0.0.1 the following way: > > > > vars.xml > > > > > > internal.xml > > > > > > > > The rest is standard configuration. > > > > Now communication Laptop-internal is UDP on port 5060 and external via > > TLS on port 5081, so I have no open port 5060 to the internet. > > > > Best regards > > Peter > > > > > > > > > > Brian West schrieb: > >> > >> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> 1st Question: Is that possible or is another solution preferrable? > >> > >> Just use FreeSWITCH with mod_portaudio. > >> > >>> 2nd Question: How can I change the amount of memory FS tries to reserve > >>> to an absolute minumum (I only have 1 call at a time). Currently it > >>> tries to reserve about 360M if I read that right. > >> > >> Thats virtual. Look at RES. > >> > >>> > >>> Best regards > >>> Peter > >> > >> Brian West > >> brian at freeswitch.org > > >> > >> -- Meet us a ClueCon! http://www.cluecon.com > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > * > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/a1890be2/attachment-0001.html From msc at freeswitch.org Tue Apr 7 23:57:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Apr 2009 23:57:37 -0700 Subject: [Freeswitch-users] need help getting ISDN talking to Cisco 3845 In-Reply-To: <200904080037.n380bKih004889@jwlab.FEITH.COM> References: <200904080037.n380bKih004889@jwlab.FEITH.COM> Message-ID: <87f2f3b90904072357n423feb79mbf206f7fd2dc962d@mail.gmail.com> John, Okay, a few things. First off, the wanpipe2.conf file has a booboo. This line is WRONG: TDMV_DCHAN = 0 For ISDN in North America you want: TDMV_DCHAN = 24 Also, I recommend changing this line: wbg1 = wanpipe2, , TDM_VOICE, Comment To this: wbg1 = wanpipe2, , TDM_VOICE_API, Comment A sample config for Sangoma wanpipeX.conf is here: http://wiki.freeswitch.org/wiki/OpenZAP#Wanpipe_mode Okay, ISDN 101: there is a "network" side and a "user" side. (Also called "terminal" or "cpe"). From what I see here you are trying to have FS be the network side and the Cisco is the user side. Assuming that this is what you want then you will need to use ozmod_libpri because the default OpenZAP PRI stack does not currently support being the network side. You will need to download and install libpri from downloads.digium.com and then you'll need to reconfigure openzap. Follow these instructions to get libpri and openzap working together: http://wiki.freeswitch.org/wiki/OpenZAP#Adding_libpri_Support And then check out this example openzap.conf.xml file for using libpri: http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples#Using_with_PRI_.28libpri_compatibility_stack.29 (Note that you don't want 'cpe' here but rather 'network'.) Now, on the Cisco side... sorry, can't help you. However, I don't see any glaring gotchas from looking at the configs. I don't see where timing is specified nor do I see where the d channel is specified. Hopefully you can confirm that those are set properly. (The cisco needs to be a "slave" to the FS clock, also called "receiving clock"; d-channel is 23 or 24 depending on how cisco numbers their channels.) Have fun! :) -MC On Tue, Apr 7, 2009 at 5:37 PM, John Wehle wrote: > Our FreeSWITCH setup has an existing T1 using RBS to talk to a digital > modem pack in a Cisco 3845. I'm interested in changing from RBS to > ISDN. I changed both sides, restart things, and see FreeSWITCH report: > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/695aac05/attachment.html From solko at gcdf.pl Wed Apr 8 00:59:15 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 09:59:15 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl> <8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> Message-ID: <49DC5953.8010000@gcdf.pl> mszlazak at aol.com pisze: > I'm not quite sure what your asking. > Are you saying that I could run the latest FS svn but in a way that uses > my "older" configuration files? If so then I don't, and don't know how > ... blush blush. > If that's the easiest thing to do then please tell me how. > Thanks. Mark. Exactly, I do it that way. For first time I gave installation prefix when configuring FS. You can stay with /usr/local/freeswich/. Now every time i call 'make current' and it does not overwrite my configuration file. In case of huge changes in modules I copy/merge my config file with the one from svn. I did not have problems with it, because developers makes good default values for new configuration options. I don't know which modules do you use, but in ones I use configuration is not changes a lot, there are new options added which does not break old one. make install do not copy configuration files for me if they are already installed, I have that on production server and all test servers. I assume this is correct behavior and I'm not the only one work like that. I looked in Makefile and it tests for config file before installing, so it does not overwrite them. Regards Szymon > > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 7 Apr 2009 10:54 pm > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > mszlazak at aol.com pisze: > >> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > >> new trunk and I have to go through vars.xml, etc changing > >> $${local_ip_v4} like you did. > >> > >> Is there a way to change $${local_ip_v4} in one place. That way one > >> wouldn't have remember all the locations that it needs to be changed? > >> > > My configuration is not updated when I compile new version and install it. Do > > you run FS with configuration path pointed to svn > > work dir? > > > >> -----Original Message----- > >> From: Peter P GMX > > >> To: freeswitch-users at lists.freeswitch.org > >> Sent: Tue, 7 Apr 2009 3:52 pm > >> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> 127.0.0.1 and memory consumption > >> > >> Thanks Brian, > >> > >> > >> > >> what I was actually looking for was to use a standard SIP soft phone > >> > >> with some additional features. > >> > >> > >> > >> I finally manged to make FS listen on 127.0.0.1 the following way: > >> > >> > >> > >> vars.xml > >> > >> > >> > >> > >> > >> internal.xml > >> > >> > >> > >> > >> > >> > >> > >> The rest is standard configuration. > >> > >> > >> > >> Now communication Laptop-internal is UDP on port 5060 and external via > >> > >> TLS on port 5081, so I have no open port 5060 to the internet. > >> > >> > >> > >> Best regards > >> > >> Peter > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> Brian West schrieb: > >> > >>> > >> > >>> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> > >> > >>>> 1st Question: Is that possible or is another solution preferrable? > >> > >>> > >> > >>> Just use FreeSWITCH with mod_portaudio. > >> > >>> > >> > >>>> 2nd Question: How can I change the amount of memory FS tries to reserve > >> > >>>> to an absolute minumum (I only have 1 call at a time). Currently it > >> > >>>> tries to reserve about 360M if I read that right. > >> > >>> > >> > >>> Thats virtual. Look at RES. > >> > >>> > >> > >>>> > >> > >>>> Best regards > >> > >>>> Peter > >> > >>> > >> > >>> Brian West > >> > >>> brian at freeswitch.org > > > ?>> > >> > >>> > >> > >>> -- Meet us a ClueCon! http://www.cluecon.com > >> > >>> > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> > >> > >>> _______________________________________________ > >> > >>> Freeswitch-users mailing list > >> > >>> Freeswitch-users at lists.freeswitch.org > > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> http://www.freeswitch.org > >> > >>> > >> > >> > >> > >> _______________________________________________ > >> > >> Freeswitch-users mailing list > >> > >> Freeswitch-users at lists.freeswitch.org > > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> ------------------------------------------------------------------------ > >> *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > >> * > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Wed Apr 8 02:27:35 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 08 Apr 2009 11:27:35 +0200 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49DA2928.9030205@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> <1239030142502-2593558.post@n2.nabble.com> <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> <49DA2928.9030205@ewetel.de> Message-ID: <49DC6E07.9000307@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Anthony, after one day running the actual trunk things looks much better than before. FS started 24h ago with 129MB VRAM and grows to 136MB VRAM by now. CPU is around 1.3% Thanks for your work! regards helmut On 06.04.2009 18:09, Helmut Kuper wrote: > Hello Anthony, > > I did a fresh checkout, compiled it, installed it into a clean directory > and will switch over to it tomorrow morning. I hope I can reuse this > existing directories: -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFJ3G4H4tZeNddg3dwRAjU7AJ0T9Fl230VfOS00Wbot3A1DTZtBUwCghFHw /CrpYYhdSGmFy+C6RxaIK1A= =yeof -----END PGP SIGNATURE----- From Prometheus001 at gmx.net Wed Apr 8 03:41:22 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 12:41:22 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl> <8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> Message-ID: <49DC7F52.3050301@gmx.net> Simply do a " make current" without a "make samples". That way the conf files in /usr/local/freeswitch/conf remain untouched. I really very seldomly update the conf files. Best regards Peter mszlazak at aol.com schrieb: > I'm not quite sure what your asking. > Are you saying that I could run the latest FS svn but in a way that > uses my "older" configuration files? If so then I don't, and don't > know how ... blush blush. > If that's the easiest thing to do then please tell me how. > Thanks. Mark. > > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 7 Apr 2009 10:54 pm > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > mszlazak at aol.com pisze: > > > I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > > > new trunk and I have to go through vars.xml, etc changing > > > $${local_ip_v4} like you did. > > > > > > Is there a way to change $${local_ip_v4} in one place. That way one > > > wouldn't have remember all the locations that it needs to be changed? > > > > > My configuration is not updated when I compile new version and install it. Do > > you run FS with configuration path pointed to svn > > work dir? > > > > > -----Original Message----- > > > From: Peter P GMX > > > > To: freeswitch-users at lists.freeswitch.org > > > Sent: Tue, 7 Apr 2009 3:52 pm > > > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > > > 127.0.0.1 and memory consumption > > > > > > Thanks Brian, > > > > > > > > > > > > what I was actually looking for was to use a standard SIP soft phone > > > > > > with some additional features. > > > > > > > > > > > > I finally manged to make FS listen on 127.0.0.1 the following way: > > > > > > > > > > > > vars.xml > > > > > > > > > > > > > > > > > > internal.xml > > > > > > > > > > > > > > > > > > > > > > > > The rest is standard configuration. > > > > > > > > > > > > Now communication Laptop-internal is UDP on port 5060 and external via > > > > > > TLS on port 5081, so I have no open port 5060 to the internet. > > > > > > > > > > > > Best regards > > > > > > Peter > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Brian West schrieb: > > > > > >> > > > > > >> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > > > > > >> > > > > > >>> 1st Question: Is that possible or is another solution preferrable? > > > > > >> > > > > > >> Just use FreeSWITCH with mod_portaudio. > > > > > >> > > > > > >>> 2nd Question: How can I change the amount of memory FS tries to reserve > > > > > >>> to an absolute minumum (I only have 1 call at a time). Currently it > > > > > >>> tries to reserve about 360M if I read that right. > > > > > >> > > > > > >> Thats virtual. Look at RES. > > > > > >> > > > > > >>> > > > > > >>> Best regards > > > > > >>> Peter > > > > > >> > > > > > >> Brian West > > > > > >> brian at freeswitch.org > > > ?>> > > > > > >> > > > > > >> -- Meet us a ClueCon! http://www.cluecon.com > > > > > >> > > > > > >> > > > > > >> > > > > > >> ------------------------------------------------------------------------ > > > > > >> > > > > > >> _______________________________________________ > > > > > >> Freeswitch-users mailing list > > > > > >> Freeswitch-users at lists.freeswitch.org > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > >> http://www.freeswitch.org > > > > > >> > > > > > > > > > > > > _______________________________________________ > > > > > > Freeswitch-users mailing list > > > > > > Freeswitch-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > ------------------------------------------------------------------------ > > > *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > > > * > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Wed Apr 8 04:10:34 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 13:10:34 +0200 Subject: [Freeswitch-users] Conference fails with speex codec In-Reply-To: <3C6448A5-1D6A-4612-80A2-644CF0BE8F88@freeswitch.org> References: <49DBDBB1.3090404@gmx.net> <87f2f3b90904071635y11d0ccd7gbfab0537cdfbb3d4@mail.gmail.com> <3C6448A5-1D6A-4612-80A2-644CF0BE8F88@freeswitch.org> Message-ID: <49DC862A.7060402@gmx.net> That was it. I installed the hd sounds and it works now. Thanks. Brian West schrieb: > Chances are he just doesn't have the 16k sound files installed. > > /b > > On Apr 7, 2009, at 6:35 PM, Michael Collins wrote: > >> >> 2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error >> Opening File >> [/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav] >> [System error : No such file or directory.] >> 2009-04-08 00:43:24 [WARNING] mod_conference.c:4799 >> conference_function() Cannot ask the user for a pin, ending >> >> >> This is curious. Do you see this error about the missing file when >> you use PCMU? >> -MC >> >> >> >> call2009-04-08 00:43:24 [NOTICE] mod_conference.c:4800 >> conference_function() Hangup >> sofia/internal/723328 at sip.mydomain.com >> >> [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Wed Apr 8 04:54:50 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 13:54:50 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> References: <49DB8B14.70400@gmx.net> <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> Message-ID: <49DC908A.9010804@gmx.net> I just wanted to know, how much memory overall is consumed by FS inkl. all Libraries (when used on a Netbook with limited memory), so RES does only show a portion of the overall RAM, FS uses incl. libraries. So I did the following: I restarted my laptop and noted the used memory. I deactivated all not needed modules in FS, started FS and noted the used memory. The difference was 24MB. When a call was present (incl. TLS/SRTP), I noted 25M. This is a really low value. Impressive!. Good job done! Best regards Peter Brian West schrieb: > >> 2nd Question: How can I change the amount of memory FS tries to reserve >> to an absolute minumum (I only have 1 call at a time). Currently it >> tries to reserve about 360M if I read that right. > > Thats virtual. Look at RES. > From solko at gcdf.pl Wed Apr 8 05:12:57 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 14:12:57 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DC908A.9010804@gmx.net> References: <49DB8B14.70400@gmx.net> <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DC908A.9010804@gmx.net> Message-ID: <49DC94C9.6000302@gcdf.pl> Peter P GMX pisze: > I just wanted to know, how much memory overall is consumed by FS inkl. > all Libraries (when used on a Netbook with limited memory), so RES does > only show a portion of the overall RAM, FS uses incl. libraries. > > So I did the following: > I restarted my laptop and noted the used memory. > I deactivated all not needed modules in FS, started FS and noted the > used memory. > The difference was 24MB. When a call was present (incl. TLS/SRTP), I > noted 25M. This is a really low value. Impressive!. > Good job done! > Do you use linux based system? Linux don't return memory once used to free, it uses it for disk buffers but it will free it when needed. So probably much part of that system was for disk buffers and are not used by FS any more. I always thought that memory allocated in libraries are included in process which is using them. Where it should be in your opinion? For external services/servers memory is not included in process but this is not the case in FS. Look at RES to know how much memory it uses. > Best regards > Peter > > Brian West schrieb: >>> 2nd Question: How can I change the amount of memory FS tries to reserve >>> to an absolute minumum (I only have 1 call at a time). Currently it >>> tries to reserve about 360M if I read that right. >> Thats virtual. Look at RES. >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 solko at gcdf.pl Wed Apr 8 05:28:11 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 14:28:11 +0200 Subject: [Freeswitch-users] how to set CF_VERBOSE_EVENTS Message-ID: <49DC985B.8080202@gcdf.pl> I track channels via mod_socket_event, I saw in source there is such flag CF_VERBOSE_EVENTS to make all channel related events contain extended data. Is it possible to set it via 'originate' or 'conference xxx dial' commands. This would ease my system. In scenario when I call user which is not registered I get only CHANNEL_DESTROY event but I cannot connect it to my command. I can monitor BACKGROUND_JOB events but in some cases (like answering) it can take time to come. Basically now I need to wait for first event of those two types (CHANNEL_ORIGINATE , BACKGROUND_JOB), if I have extended data in all events I can ignore BACKGROUND_JOB. Regards Szymon From dujinfang at gmail.com Wed Apr 8 07:15:11 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 8 Apr 2009 22:15:11 +0800 Subject: [Freeswitch-users] how to set CF_VERBOSE_EVENTS In-Reply-To: <49DC985B.8080202@gcdf.pl> References: <49DC985B.8080202@gcdf.pl> Message-ID: <5E75EFE2-1653-444C-BA20-6C0BD6335546@gmail.com> There is a dp_tools verbose_events can set that flag, you may try to transfer into a dialplan or use the inline dialplan try this, not tested. > originate sofia/gateway/my_gw/user at domain.com 'verbose_events,playback:foo.wav,echo' inline http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_InlineDialplan On Apr 8, 2009, at 8:28 PM, Szymon Olko wrote: > I track channels via mod_socket_event, I saw in source there is such > flag CF_VERBOSE_EVENTS to make all channel related events > contain extended data. Is it possible to set it via 'originate' or > 'conference xxx dial' commands. > > This would ease my system. In scenario when I call user which is not > registered I get only CHANNEL_DESTROY event but I cannot > connect it to my command. I can monitor BACKGROUND_JOB events but in > some cases (like answering) it can take time to come. > > Basically now I need to wait for first event of those two types > (CHANNEL_ORIGINATE , BACKGROUND_JOB), if I have extended data in > all events I can ignore BACKGROUND_JOB. > > Regards > > Szymon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 carlos.talbot at gmail.com Wed Apr 8 07:38:10 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 8 Apr 2009 09:38:10 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <5800526b0904031429s3b1deb4do13ecf3335e18949a@mail.gmail.com> References: <7.0.1.0.2.20090403120452.024273d8@fredshack.com> <5800526b0904031429s3b1deb4do13ecf3335e18949a@mail.gmail.com> Message-ID: <5800526b0904080738t78a10216o7fd78df9f97bad71@mail.gmail.com> Here's the first draft: http://wiki.freeswitch.org/wiki/OpenWrt Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/f9bf968f/attachment.html From tvietduc at yahoo.com Tue Apr 7 19:49:42 2009 From: tvietduc at yahoo.com (to vietduc) Date: Tue, 7 Apr 2009 19:49:42 -0700 (PDT) Subject: [Freeswitch-users] Help about Conference and it's member_id Message-ID: <199683.40252.qm@web38107.mail.mud.yahoo.com> ?Hi! ?I wonder how to get out the member_id of a member in a conference room? Currently, i found that it is increasing 1 each time, so the first one enter the conference room, his/her member_id would be 1, and the second is 2. Sadly, when i close a conference room (all members leave) and re-open it later, the member_id will be increased since the last time instead of begining at 1 as the first time (said, it may be 3, 4 and go on). Why is that (as i think a usual way is that if there is 3 members in a conference room, their member_id should be 1,2,3 or 0,1,2) and is there anyway to get conference's member_id of a member programatically? ?Thanks in advance! ?Duc To From mchlmll at gmail.com Wed Apr 8 03:34:02 2009 From: mchlmll at gmail.com (Michele M) Date: Wed, 8 Apr 2009 12:34:02 +0200 Subject: [Freeswitch-users] How to design my project ? Message-ID: Hi there, I'm quite a newbie about freeswitch. I have an application (IVR) that needs to have endpoints SIP to register,answer the calls and transfer them to the right phones.(I( have my own SIP server).Moreover it needs also a ASR/TTS API' set to communicate with my ASR/TTS engine ( just for example let's assume it is Cepstral). I'd wouldn't want to have freeswitch running and communicate with it to accomplish that but just to use the libfreeswitch library embedded. As I don't know that much about freeswitch can it be done? or just I need to have freeswitch running as a must? Can somebody point me to the right place where to find example of using library embedded (best examples for what I'm trying to do) as I have not found that many? Thanks in advance Miki -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/20fc3e62/attachment.html From wiltingtree at gmail.com Wed Apr 8 08:18:55 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Wed, 8 Apr 2009 11:18:55 -0400 Subject: [Freeswitch-users] Two or more simultaneous calls not Message-ID: Thanks for the response Traun. The version of Python is 2.4.3, and I didn't build it myself, I installed it with yum. The version of Red Hat is 4.1.2-41. "import threading" works fine, so I don't think it's a Python threading issue. The FreeSWITCH version I installed is the freeswitch-1.0.3.tar.gz located at files.freeswitch.org. I didn't make any major changes to the configuration; I enabled Python and set-up the SIP profile, directory and dialplan. No other changes. Any other help would be appreciated, since I really don't know where to look. Thanks, Adam >Date: Wed, 8 Apr 2009 06:36:27 +0430 >From: Traun Leyden >Subject: Re: [Freeswitch-users] Two or more simultaneous calls not > working >To: freeswitch-users at lists.freeswitch.org >Message-ID: > >Content-Type: text/plain; charset="iso-8859-1" > >> >> >> Hello, >> >> I wrote an application using FreeSWITCH version 1.0.3, with mod_python and >> a >> 64 bit box on Red Hat. >> The app works fine when one person dials in, but when a second person dials >> in, the first call stops and waits until the second call is finished. It's >> really strange - if the first call is right in the middle of playing a >> prompt, it will just stop, and there will be dead air. As soon as the 2nd >> call hangs up, the prompt for the first call starts playing right where it >> left off. >> >> I previously had FreeSWITCH installed on a 32 bit CentOS box, and this was >> not happening. >> >> Does anybody have any idea what the cause of this could be? >> >> Thanks! > > >I'm running mod_python fine with 64-bit (Ubuntu 8) and fs svn 12793, and >have not seen that problem. Which version of python? Did you build it >yourself or was it from a package? If its from a package, please provide >the version of Red Hat you are using. > >One possible cause is that python was not compiled with multi-threading >support. I don't know how to check that however .. googled around and >didn't find anything. If you fire up >your python interpreter and type "import threading" do you get an error? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/f5cf95c7/attachment.html From solko at gcdf.pl Wed Apr 8 09:05:42 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 18:05:42 +0200 Subject: [Freeswitch-users] Help about Conference and it's member_id In-Reply-To: <199683.40252.qm@web38107.mail.mud.yahoo.com> References: <199683.40252.qm@web38107.mail.mud.yahoo.com> Message-ID: <49DCCB56.6090502@gcdf.pl> to vietduc pisze: > Hi! > I wonder how to get out the member_id of a member in a conference room? Currently, i found that it is increasing 1 each time, so the first one enter the conference room, his/her member_id would be 1, and the second is 2. Sadly, when i close a conference room (all members leave) and re-open it later, the member_id will be increased since the last time instead of begining at 1 as the first time (said, it may be 3, 4 and go on). Why is that (as i think a usual way is that if there is 3 members in a conference room, their member_id should be 1,2,3 or 0,1,2) and is there anyway to get conference's member_id of a member programatically? > Thanks in advance! > Duc To > First of all members id is increased in FS instance. So you will never get the same id again, unless you reload FS (maybe mod_conference reload also). There is a good reason for that so once you have a member it id stays the same no matter in how many conferences it pass through. All conference commands needs that id so you would be lost if that would change all the time. Yuu did not wrote in which language and how you need it, I'm using mod_event_socket and there listen for CUSTOME_EVENT subevent is conference:maintanace, actions ADDED, DELETED. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 andy at fabulous4.co.uk Wed Apr 8 09:18:23 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Wed, 8 Apr 2009 17:18:23 +0100 Subject: [Freeswitch-users] Using recordFile with Icecast - looses the end of the call Message-ID: <2F161DA684214D6487B8F9711563C62F@wsandy> Hi, I have mod_shout installed and I'm using session.recordFile to capture the audio in a call. When I specify a local file mp3 or wav the audio is captured fine. However, I'm using an icecast server to manage the audio for me and when I specify a remote mp3 location(shout://myserver.com/myaudio.mp3) the end of the call is missing off the resultant mp3 file. A wild shot in the dark I know but does anyone have any experience of this and how it might be resolved? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/a2c3cb19/attachment.html From mszlazak at aol.com Wed Apr 8 09:18:21 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 08 Apr 2009 12:18:21 -0400 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DC5953.8010000@gcdf.pl> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> Message-ID: <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> OK , you're SVN updating on a Linux system but I'm using Windows. The very few times I tried with Tortoise SVN I ran into problems were it would fail because of some path not being present or some strange symbol in a file or something else. Since I'm not experienced enough and don't always have the time, I gave up on this approach and just start over again in a different folder then reconfigure the updated FS and transfer files from an older FS. Yup, it sucks. Thanks.? -----Original Message----- From: Szymon Olko To: freeswitch-users at lists.freeswitch.org Sent: Wed, 8 Apr 2009 12:59 am Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption mszlazak at aol.com pisze: > I'm not quite sure what your asking. > Are you saying that I could run the latest FS svn but in a way that uses > my "older" configuration files? If so then I don't, and don't know how > ... blush blush. > If that's the easiest thing to do then please tell me how. > Thanks. Mark. Exactly, I do it that way. For first time I gave installation prefix when configuring FS. You can stay with /usr/local/freeswich/. Now every time i call 'make current' and it does not overwrite my configuration file. In case of huge changes in modules I copy/merge my config file with the one from svn. I did not have problems with it, because developers makes good default values for new configuration options. I don't know which modules do you use, but in ones I use configuration is not changes a lot, there are new options added which does not break old one. make install do not copy configuration files for me if they are already installed, I have that on production server and all test servers. I assume this is correct behavior and I'm not the only one work like that. I looked in Makefile and it tests for config file before installing, so it does not overwrite them. Regards Szymon > > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 7 Apr 2009 10:54 pm > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > mszlazak at aol.com pisze: > >> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > >> new trunk and I have to go through vars.xml, etc changing > >> $${local_ip_v4} like you did. > >> > >> Is there a way to change $${local_ip_v4} in one place. That way one > >> wouldn't have remember all the locations that it needs to be changed? > >> > > My configuration is not updated when I compile new version and install it. Do > > you run FS with configuration path pointed to svn > > work dir? > > > >> -----Original Message----- > >> From: Peter P GMX > > >> To: freeswitch-users at lists.freeswitch.org > >> Sent: Tue, 7 Apr 2009 3:52 pm > >> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> 127.0.0.1 and memory consumption > >> > >> Thanks Brian, > >> > >> > >> > >> what I was actually looking for was to use a standard SIP soft phone > >> > >> with some additional features. > >> > >> > >> > >> I finally manged to make FS listen on 127.0.0.1 the following way: > >> > >> > >> > >> vars.xml > >> > >> > >> > >> > >> > >> internal.xml > >> > >> > >> > >> > >> > >> > >> > >> The rest is standard configuration. > >> > >> > >> > >> Now communication Laptop-internal is UDP on port 5060 and external via > >> > >> TLS on port 5081, so I have no open port 5060 to the internet. > >> > >> > >> > >> Best regards > >> > >> Peter > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> Brian West schrieb: > >> > >>> > >> > >>> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> > >> > >>>> 1st Question: Is that possible or is another solution preferrable? > >> > >>> > >> > >>> Just use FreeSWITCH with mod_portaudio. > >> > >>> > >> > >>>> 2nd Question: How can I change the amount of memory FS tries to reserve > >> > >>>> to an absolute minumum (I only have 1 call at a time). Currently it > >> > >>>> tries to reserve about 360M if I read that right. > >> > >>> > >> > >>> Thats virtual. Look at RES. > >> > >>> > >> > >>>> > >> > >>>> Best regards > >> > >>>> Peter > >> > >>> > >> > >>> Brian West > >> > >>> brian at freeswitch.org > > > ?>> > >> > >>> > >> > >>> -- Meet us a ClueCon! http://www.cluecon.com > >> > >>> > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> > >> > >>> _______________________________________________ > >> > >>> Freeswitch-users mailing list > >> > >>> Freeswitch-users at lists.freeswitch.org > > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> http://www.freeswitch.org > >> > >>> > >> > >> > >> > >> _______________________________________________ > >> > >> Freeswitch-users mailing list > >> > >> Freeswitch-users at lists.freeswitch.org > > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> ------------------------------------------------------------------------ > >> *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > >> * > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/93853ff1/attachment-0001.html From Prometheus001 at gmx.net Wed Apr 8 09:52:30 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 18:52:30 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DC94C9.6000302@gcdf.pl> References: <49DB8B14.70400@gmx.net> <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DC908A.9010804@gmx.net> <49DC94C9.6000302@gcdf.pl> Message-ID: <49DCD64E.4040104@gmx.net> It's linux, yes. The way I got around the problem that memory may not be freed is: * to reboot the system. * look for used memory * start FS * look for used memory * calculate the difference That way it showed 24-25M which I can understand. Best regards Peter Szymon Olko schrieb: > Peter P GMX pisze: > >> I just wanted to know, how much memory overall is consumed by FS inkl. >> all Libraries (when used on a Netbook with limited memory), so RES does >> only show a portion of the overall RAM, FS uses incl. libraries. >> >> So I did the following: >> I restarted my laptop and noted the used memory. >> I deactivated all not needed modules in FS, started FS and noted the >> used memory. >> The difference was 24MB. When a call was present (incl. TLS/SRTP), I >> noted 25M. This is a really low value. Impressive!. >> Good job done! >> >> > Do you use linux based system? Linux don't return memory once used to free, it uses it for disk buffers but it will free it when > needed. So probably much part of that system was for disk buffers and are not used by FS any more. > > I always thought that memory allocated in libraries are included in process which is using them. Where it should be in your > opinion? For external services/servers memory is not included in process but this is not the case in FS. Look at RES to know how > much memory it uses. > > >> Best regards >> Peter >> >> Brian West schrieb: >> >>>> 2nd Question: How can I change the amount of memory FS tries to reserve >>>> to an absolute minumum (I only have 1 call at a time). Currently it >>>> tries to reserve about 360M if I read that right. >>>> >>> Thats virtual. Look at RES. >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 gmaruzz at celliax.org Wed Apr 8 09:56:37 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 8 Apr 2009 18:56:37 +0200 Subject: [Freeswitch-users] How to design my project ? In-Reply-To: References: Message-ID: <7b197bef0904080956h190a0bb1v6fda1c6965931390@mail.gmail.com> Ciao Michele, as a start is definitely better (and more gratifying) that you runs FreeSWITCH. Then, if (and only if) there is a compelling reason that justify the amount of time needed to develop a standalone application, go for it. Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Apr 8, 2009 at 12:34 PM, Michele M wrote: > Hi there, > > I'm quite a newbie about freeswitch. I have an? application? (IVR) that > needs to have endpoints SIP to register,answer the calls and transfer them > to the right phones.(I( have my own SIP server).Moreover it needs also a > ASR/TTS API' set? to communicate with my ASR/TTS engine ( just for example > let's assume it is Cepstral). I'd wouldn't want to have freeswitch running > and communicate with it to accomplish that but just to use the libfreeswitch > library embedded. As I don't know that much about freeswitch can it be done? > or just I need to have freeswitch running as a must? Can somebody point me > to the right place where to find example of using library embedded (best > examples for what I'm trying to do) as I have not found that many? > > Thanks in advance > > Miki > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 solko at gcdf.pl Wed Apr 8 10:11:29 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 19:11:29 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DCD64E.4040104@gmx.net> References: <49DB8B14.70400@gmx.net> <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DC908A.9010804@gmx.net> <49DC94C9.6000302@gcdf.pl> <49DCD64E.4040104@gmx.net> Message-ID: <49DCDAC1.70408@gcdf.pl> Peter P GMX pisze: > It's linux, yes. > The way I got around the problem that memory may not be freed is: > > * to reboot the system. > * look for used memory > * start FS > * look for used memory > * calculate the difference > > That way it showed 24-25M which I can understand. > I meant that FS can use less memory now they 24-25 M, what is RES shows is exactly that value. In those 24-25 M are buffers for files which now are not needed for FS and kernel handles that memory. It is show as used but is not used by FS. Kernel uses it and will free it when there will be lack of memory. Always look at RES value if you want to know FS consumption. Szymon > Best regards > Peter > > Szymon Olko schrieb: >> Peter P GMX pisze: >> >>> I just wanted to know, how much memory overall is consumed by FS inkl. >>> all Libraries (when used on a Netbook with limited memory), so RES does >>> only show a portion of the overall RAM, FS uses incl. libraries. >>> >>> So I did the following: >>> I restarted my laptop and noted the used memory. >>> I deactivated all not needed modules in FS, started FS and noted the >>> used memory. >>> The difference was 24MB. When a call was present (incl. TLS/SRTP), I >>> noted 25M. This is a really low value. Impressive!. >>> Good job done! >>> >>> >> Do you use linux based system? Linux don't return memory once used to free, it uses it for disk buffers but it will free it when >> needed. So probably much part of that system was for disk buffers and are not used by FS any more. >> >> I always thought that memory allocated in libraries are included in process which is using them. Where it should be in your >> opinion? For external services/servers memory is not included in process but this is not the case in FS. Look at RES to know how >> much memory it uses. >> >> >>> Best regards >>> Peter >>> >>> Brian West schrieb: >>> >>>>> 2nd Question: How can I change the amount of memory FS tries to reserve >>>>> to an absolute minumum (I only have 1 call at a time). Currently it >>>>> tries to reserve about 360M if I read that right. >>>>> >>>> Thats virtual. Look at RES. >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 solko at gcdf.pl Wed Apr 8 10:36:37 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 19:36:37 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> Message-ID: <49DCE0A5.6080905@gcdf.pl> mszlazak at aol.com pisze: > OK , you're SVN updating on a Linux system but I'm using Windows. The > very few times I tried with Tortoise SVN I ran into problems were it > would fail because of some path not being present or some strange symbol > in a file or something else. Since I'm not experienced enough and don't > always have the time, I gave up on this approach and just start over > again in a different folder then reconfigure the updated FS and transfer > files from an older FS. Yup, it sucks. > Yes I'm linux user. If you have problems with svn update then you can do your way, make fresh checkout every time. After CO copy modules.conf and build new version, just copy old config files to installation directory if it is always different one. That's why I hate gui tools for things like full svn update. > > > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, 8 Apr 2009 12:59 am > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > mszlazak at aol.com pisze: > >> I'm not quite sure what your asking. > >> Are you saying that I could run the latest FS svn but in a way that uses > >> my "older" configuration files? If so then I don't, and don't know how > >> ... blush blush. > >> If that's the easiest thing to do then please tell me how. > >> Thanks. Mark. > > > > Exactly, I do it that way. > > For first time I gave installation prefix when configuring FS. You can stay with > > /usr/local/freeswich/. > > Now every time i call 'make current' and it does not overwrite my configuration > > file. > > > > In case of huge changes in modules I copy/merge my config file with the one from > > svn. I did not have problems with it, because > > developers makes good default values for new configuration options. > > > > I don't know which modules do you use, but in ones I use configuration is not > > changes a lot, there are new options added which > > does not break old one. > > > > make install do not copy configuration files for me if they are already > > installed, I have that on production server and all test > > servers. I assume this is correct behavior and I'm not the only one work like > > that. > > I looked in Makefile and it tests for config file before installing, so it does > > not overwrite them. > > > > Regards > > Szymon > >> > >> -----Original Message----- > >> From: Szymon Olko > > >> To: freeswitch-users at lists.freeswitch.org > >> Sent: Tue, 7 Apr 2009 10:54 pm > >> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> 127.0.0.1 and memory consumption > >> > >> mszlazak at aol.com > pisze: > >> > >>> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > >> > >>> new trunk and I have to go through vars.xml, etc changing > >> > >>> $${local_ip_v4} like you did. > >> > >>> > >> > >>> Is there a way to change $${local_ip_v4} in one place. That way one > >> > >>> wouldn't have remember all the locations that it needs to be changed? > >> > >>> > >> > >> My configuration is not updated when I compile new version and install it. Do > >> > >> you run FS with configuration path pointed to svn > >> > >> work dir? > >> > >> > >> > >>> -----Original Message----- > >> > >>> From: Peter P GMX >> > >> > >>> To: freeswitch-users at lists.freeswitch.org > > >> > >>> Sent: Tue, 7 Apr 2009 3:52 pm > >> > >>> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> > >>> 127.0.0.1 and memory consumption > >> > >>> > >> > >>> Thanks Brian, > >> > >>> > >> > >>> > >> > >>> > >> > >>> what I was actually looking for was to use a standard SIP soft phone > >> > >>> > >> > >>> with some additional features. > >> > >>> > >> > >>> > >> > >>> > >> > >>> I finally manged to make FS listen on 127.0.0.1 the following way: > >> > >>> > >> > >>> > >> > >>> > >> > >>> vars.xml > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> internal.xml > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> The rest is standard configuration. > >> > >>> > >> > >>> > >> > >>> > >> > >>> Now communication Laptop-internal is UDP on port 5060 and external via > >> > >>> > >> > >>> TLS on port 5081, so I have no open port 5060 to the internet. > >> > >>> > >> > >>> > >> > >>> > >> > >>> Best regards > >> > >>> > >> > >>> Peter > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> Brian West schrieb: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> 1st Question: Is that possible or is another solution preferrable? > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Just use FreeSWITCH with mod_portaudio. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> 2nd Question: How can I change the amount of memory FS tries to reserve > >> > >>> > >> > >>>>> to an absolute minumum (I only have 1 call at a time). Currently it > >> > >>> > >> > >>>>> tries to reserve about 360M if I read that right. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Thats virtual. Look at RES. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>>> Best regards > >> > >>> > >> > >>>>> Peter > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Brian West > >> > >>> > >> > >>>> brian at freeswitch.org > > > ?>> > > ?> > >> > >> ?>?>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> -- Meet us a ClueCon! http://www.cluecon.com > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> ------------------------------------------------------------------------ > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> _______________________________________________ > >> > >>> > >> > >>>> Freeswitch-users mailing list > >> > >>> > >> > >>>> Freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>>> http://www.freeswitch.org > >> > >>> > >> > >>>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> _______________________________________________ > >> > >>> > >> > >>> Freeswitch-users mailing list > >> > >>> > >> > >>> Freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>> http://www.freeswitch.org > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > >> > >>> * > > > >> > >>> > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> > >> > >>> _______________________________________________ > >> > >>> Freeswitch-users mailing list > >> > >>> Freeswitch-users at lists.freeswitch.org > > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> _______________________________________________ > >> > >> Freeswitch-users mailing list > >> > >> Freeswitch-users at lists.freeswitch.org > > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> ------------------------------------------------------------------------ > >> New Deals on Dell Netbooks - Now starting at $299 > >> > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 8 10:44:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Apr 2009 12:44:04 -0500 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DCE0A5.6080905@gcdf.pl> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> Message-ID: <3BBB82A7-FA78-4FDB-A44B-672B942C66E5@freeswitch.org> You know if you keep doing a fresh checkout every single time then you are wasting bandwidth... if its your only choice then do that but I highly recommend you learn to use the tools properly. Our bandwidth is kindly provided by Bandwidth.com and I would hate to just waste it for no reason.... btw don't forget to register for Cluecon its quickly approaching. /b On Apr 8, 2009, at 12:36 PM, Szymon Olko wrote: > Yes I'm linux user. > If you have problems with svn update then you can do your way, make > fresh checkout every time. After CO copy modules.conf and > build new version, just copy old config files to installation > directory if it is always different one. > > That's why I hate gui tools for things like full svn update. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/db4ed547/attachment-0001.html From mszlazak at aol.com Wed Apr 8 10:58:31 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 08 Apr 2009 13:58:31 -0400 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DCE0A5.6080905@gcdf.pl> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl><8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> Message-ID: <8CB868AC2870B10-8D4-75@MBLK-M41.sysops.aol.com> Na jaki? czas, b?d? uczy? si?, jak radzi? sobie z Tortoise SVN b??d?w. M?j polski nie jest zbyt dobre, ale dzi?kuj?. Google pomaga. -----Original Message----- From: Szymon Olko To: freeswitch-users at lists.freeswitch.org Sent: Wed, 8 Apr 2009 10:36 am Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption mszlazak at aol.com pisze: > OK , you're SVN updating on a Linux system but I'm using Windows. The > very few times I tried with Tortoise SVN I ran into problems were it > would fail because of some path not being present or some strange symbol > in a file or something else. Since I'm not experienced enough and don't > always have the time, I gave up on this approach and just start over > again in a different folder then reconfigure the updated FS and transfer > files from an older FS. Yup, it sucks. > Yes I'm linux user. If you have problems with svn update then you can do your way, make fresh checkout every time. After CO copy modules.conf and build new version, just copy old config files to installation directory if it is always different one. That's why I hate gui tools for things like full svn update. > > > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, 8 Apr 2009 12:59 am > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > =0 A> mszlazak at aol.com pisze: > >> I'm not quite sure what your asking. > >> Are you saying that I could run the latest FS svn but in a way that uses > >> my "older" configuration files? If so then I don't, and don't know how > >> ... blush blush. > >> If that's the easiest thing to do then please tell me how. > >> Thanks. Mark. > > > > Exactly, I do it that way. > > For first time I gave installation prefix when configuring FS. You can stay with > > /usr/local/freeswich/. > > Now every time i call 'make current' and it does not overwrite my configuration > > file. > > > > In case of huge changes in modules I copy/merge my config file with the one from > > svn. I did not have problems with it, because > > developers makes good default values for new configuration options. > > > > I don't know which modules do you use, but in ones I use configuration is not > > changes a lot, there are new options added which > > does not break old one. > > > > make install do not copy configuration files for me if they are already > > installed, I have that on production server and all test > > servers. I assume this is correct behavior and I'm not the only one work like > > that. > > I looked in Makefile and it tests for config file before installing, so it does > > not overwrite them. > > > > Regards > > Szymon > >>=2 0 > >> -----Original Message----- > >> From: Szymon Olko > > >> To: freeswitch-users at lists.freeswitch.org > >> Sent: Tue, 7 Apr 2009 10:54 pm > >> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> 127.0.0.1 and memory consumption > >> > >> mszlazak at aol.com > pisze: > >> > >>> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > >> > >>> new trunk and I have to go through vars.xml, etc changing > >> > >>> $${local_ip_v4} like you did. > >> > >>> > >> > >>> Is there a way to change $${local_ip_v4} in one place. That way one > >> > >>> wouldn't have remember all the locations that it needs to be changed? > >> > >>> > >> > >> My configuration is not updated when I compile new version and install it. Do > >> > >> you run FS with configuration path pointed to svn > >> > >> work dir? > >> > >> > >> > >>> -----Original Message----- > >> > >>> From: Peter P GMX >> > >> > >>> To: freeswitch-users at lists.freeswitch.org > > >> > >>> Sent: Tue, 7 Apr 2009 3:52 pm > >> > >>> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> > >>> 127.0.0.1 and memory consumption > >> > >>> > >> > >>> Thanks Brian, > >> > >>> > >> > >>> > >> > >>> > >> > >>> what I was actually looking for was to use a standard SIP soft phone > >> > >>> > >> > >>> with some additional features. > >> > >>> > >> > >>> > >> > >>> > >> > >>> I finally manged to make FS listen on 127.0.0.1 the following way: > >> > >>> > >> > >>> > >> > >>> > >> > >>> vars.xml > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> internal.xml > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> The rest is standard configuration. > >> > >>> > >> > >>> > >> > >>> > >> > >>> Now communication Laptop-internal is UDP on port 5060 and external via > >> > >>> > >> > >>> TLS on port 5081, so I have no open port 5060 to the internet. > >> > >>> > >> > >>> > >> > >>> > >> > >>> Best regards > >> > >>> > >> > >>> Peter > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> Brian West schrieb: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> 1st Question: Is that possible or is another solution preferrable? > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Just use FreeSWITCH with mod_portaudio. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> 2nd Question: How can I change the amount of memory FS tries to reserve > >> > >>> > >> > >>>>> to an absolute minumum (I only have 1 call at a time). Currently it > >> > >>> > >> > >>>>> tries to reserve about 360M if I read that right. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Thats virtual. Look at RES. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>>> Best regards > >> > >>> > >> > >>>>> Peter > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Brian West > >> > >>> > >> > >>>> brian at freeswitch.org > > > ?>> > > ?> > >> > >> ?>?>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> -- Meet us a ClueCon! http://www.cluecon.com > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> ------------------------------------------------------------------------ > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> _______________________________________________ > >> > >>> > >> > >>>> Freeswitch-users mailing list > >> > >>> > >> > >>>> Freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswi tch-users > >> > >>> > >> > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>>> http://www.freeswitch.org > >> > >>> > >> > >>>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> _______________________________________________ > >> > >>> > >> > >>> Freeswitch-users mailing list > >> > >>> > >> > >>> Freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>> http://www.freeswitch.org > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > >> > >>> * > > > >> > >>> > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> > >> > >>> _______________________________________________ > >> > >>> Freeswitch-users mailing list > >> > >>> Freeswitch-users at lists.freeswitch.org > > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> _______________________________________________ > >> > >> Freeswitch-users mailing list > >> > >> Freeswitch-users at lists.freeswitch.org > > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> ------------------------------------------------------------------------ > >> New Deals on Dell Netbooks - Now starting at $299 > >> > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/fr eeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/5b019a7e/attachment-0001.html From gkuri at ieee.org Wed Apr 8 11:08:53 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Wed, 08 Apr 2009 11:08:53 -0700 Subject: [Freeswitch-users] rtp/one way audio problem Message-ID: <49DCE835.8030206@ieee.org> We're seeing occasional one way audio issues for international calls going out to one of several carriers. On roughly 2 out of 5 calls outbound, there is no audio on the the calling party's side, however the called party indicates they can hear the calling party perfectly well. NAT is not involved anywhere on our side. Sniffing the traffic shows the rtp stream tries to start, coming in from the carrier, but then stops, which is probably why there's no audio on the calling party's side. There is however, rtp going out from us to the carrier, which is probably why the called party hears the calling party OK. I enabled proxy_media=true and that seems to have fixed the problem (or the problem coincidentally stopped), so now I'm beginning to wonder if we're hitting any of the goofy Sonus bugs described here ... http://wiki.freeswitch.org/wiki/RTP_Issues I know my carrier uses a Cisco, but I also know their routes are transit, not direct routes, so I have no idea what other softswitches the rtp is going through before it finally hits the PSTN in whatever country is being called. Is there anyway to know if there's a Sonus in the media stream somewhere? Do the Cisco's do anything goofy that I need to be aware of as well? I can turn off proxy_media and run a pcapsipdump if that will help? As a side note, we have absolutely no problem terminating domestic calls via any of our domestic carriers/CLECs. This problem seems to be only plagued with International calling. Thanks, Gabe From solko at gcdf.pl Wed Apr 8 11:14:07 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 20:14:07 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <3BBB82A7-FA78-4FDB-A44B-672B942C66E5@freeswitch.org> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> <3BBB82A7-FA78-4FDB-A44B-672B942C66E5@freeswitch.org> Message-ID: <49DCE96F.5070500@gcdf.pl> Brian West pisze: > You know if you keep doing a fresh checkout every single time then you > are wasting bandwidth... if its your only choice then do that but I > highly recommend you learn to use the tools properly. Our bandwidth is > kindly provided by Bandwidth.com and I would hate to just waste it for > no reason.... btw don't forget to register for Cluecon its quickly > approaching. > > /b > Your right about bandwidth, I use svn in console and never had problems that cannot be fixed. Those gui tools they try to be to intelligent. I thought there was console svn tool for windows. Regarding Cluecon, I would like to meet you all there, but in this year it's to expensive and too far for me. Szymon > On Apr 8, 2009, at 12:36 PM, Szymon Olko wrote: > >> Yes I'm linux user. >> If you have problems with svn update then you can do your way, make >> fresh checkout every time. After CO copy modules.conf and >> build new version, just copy old config files to installation >> directory if it is always different one. >> >> That's why I hate gui tools for things like full svn update. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 8 11:23:16 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Apr 2009 13:23:16 -0500 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DCE96F.5070500@gcdf.pl> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> <3BBB82A7-FA78-4FDB-A44B-672B942C66E5@freeswitch.org> <49DCE96F.5070500@gcdf.pl> Message-ID: Where are you? /b On Apr 8, 2009, at 1:14 PM, Szymon Olko wrote: > Regarding Cluecon, I would like to meet you all there, but in this > year it's to expensive and too far for me. > > Szymon Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/c9ebb066/attachment.html From brian at freeswitch.org Wed Apr 8 11:23:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Apr 2009 13:23:50 -0500 Subject: [Freeswitch-users] rtp/one way audio problem In-Reply-To: <49DCE835.8030206@ieee.org> References: <49DCE835.8030206@ieee.org> Message-ID: <1E09FF26-77A5-4D4B-8ACF-76909B393E19@freeswitch.org> Do you have any reason to be doing proxy media? /b On Apr 8, 2009, at 1:08 PM, Gabriel Kuri wrote: > I can turn off proxy_media and run a pcapsipdump if that will help? Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/2f6cb535/attachment.html From gkuri at ieee.org Wed Apr 8 11:30:26 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Wed, 08 Apr 2009 11:30:26 -0700 Subject: [Freeswitch-users] rtp/one way audio problem In-Reply-To: <1E09FF26-77A5-4D4B-8ACF-76909B393E19@freeswitch.org> References: <49DCE835.8030206@ieee.org> <1E09FF26-77A5-4D4B-8ACF-76909B393E19@freeswitch.org> Message-ID: <49DCED42.6070606@ieee.org> Brian West wrote: > Do you have any reason to be doing proxy media? no, not other than to fix the one way audio issue :) I'd rather leave proxy_media off. Gabe From solko at gcdf.pl Wed Apr 8 11:35:13 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 20:35:13 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> <3BBB82A7-FA78-4FDB-A44B-672B942C66E5@freeswitch.org> <49DCE96F.5070500@gcdf.pl> Message-ID: <49DCEE61.8090802@gcdf.pl> Brian West pisze: > Where are you? > Poland, Wroc?aw. http://maps.google.pl/maps?f=q&source=s_q&hl=pl&geocode=&q=poland,+wroc%C5%82aw&sll=52.025459,19.204102&sspn=6.979078,18.017578&ie=UTF8&ll=51.107833,17.038422&spn=0.222023,0.563049&z=11 > /b > > On Apr 8, 2009, at 1:14 PM, Szymon Olko wrote: > >> Regarding Cluecon, I would like to meet you all there, but in this >> year it's to expensive and too far for me. >> >> Szymon > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 can_man at gmx.de Wed Apr 8 14:24:39 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Wed, 08 Apr 2009 23:24:39 +0200 Subject: [Freeswitch-users] speex can't find OggS header Message-ID: <20090408212439.268950@gmx.net> Hello everyone, I am trying to get a Java Sip client working with speex/16000. FS sets the codec correctly and then starts sending packets to my client: 2009-04-08 21:46:34 [DEBUG] sofia_glue.c:2732 sofia_glue_negotiate_sdp() Audio Codec Compare [speex:100:16000:0]/[SPEEX:99:16000:20] 2009-04-08 21:46:34 [DEBUG] sofia_glue.c:1857 sofia_glue_tech_set_codec() Set Codec sofia/external5090/puli at 97.101.59.118:5090 SPEEX/16000 20 ms 320 samples When the packets arrive jspeex can't decode them and I started to look at them manually to find out what the problem is. The payload of each RTP packet is 42 bytes and when looking for the "OggS" header I can't find it. Or is the ogg header not needed? Jspeex looks for it and as it can't find it, it stops decoding. Is FS sending one frame per packet? Thank you very much for your help. Best wishes, Phil Ps: Wireshark tells me the following for a sample package: Real-Time Transport Protocol Setup Method: SDP 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: speex (100) Sequence number: 9365 Extended sequence number: 74901 Timestamp: 23680 Synchronization Source identifier: 0x004a235c (4858716) Payload: 2C6679456EE347FFD0A3D55B133771A9A100DB639F5B8ED2... Payload is 42 bytes. -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01 From tleyden at branchcut.com Wed Apr 8 15:06:08 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Thu, 9 Apr 2009 02:36:08 +0430 Subject: [Freeswitch-users] Two or more simultaneous calls not Message-ID: Hi Adam, I'm stumped .. I guess you could try the following: * Try with the trunk version of freeswitch. I don't think it will matter, but just in case * Try to simulate the same test with a Lua script. Do you see the same problem? If those don't turn up anything, then the next logical step would be to start adding printf() statements in the mod_python code and find out where it is getting stuck. In particular around the parts where it swaps the threadstate in and out. I might be able to create a patch for you, but try those other tests first. HTH, Traun > > Thanks for the response Traun. The version of Python is 2.4.3, and I > didn't > build it myself, I installed it with yum. > The version of Red Hat is 4.1.2-41. > "import threading" works fine, so I don't think it's a Python threading > issue. > The FreeSWITCH version I installed is the > freeswitch-1.0.3.tar.gz< > http://files.freeswitch.org/freeswitch-1.0.3.tar.gz> > located > at files.freeswitch.org. > I didn't make any major changes to the configuration; I enabled Python and > set-up the SIP profile, directory and dialplan. No other changes. > Any other help would be appreciated, since I really don't know where to > look. > > Thanks, > Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/15f8d417/attachment-0001.html From john at feith.com Wed Apr 8 15:19:00 2009 From: john at feith.com (John Wehle) Date: Wed, 8 Apr 2009 18:19:00 -0400 (EDT) Subject: [Freeswitch-users] need help getting ISDN talking to Cisco 3845 Message-ID: <200904082219.n38MJ0ld006139@jwlab.FEITH.COM> > Okay, a few things. First off, the wanpipe2.conf file has a booboo. Don't think so. > This line is WRONG: > TDMV_DCHAN = 0 Not exactly. My understanding is you can use either: wanpipeX.conf: TDMV_DCHAN = 0 zaptel.conf: dchan = 24 (or in our case 48 since it's the second span) which means use zaptel to handle the d-channel hdlc or wanpipeX.conf: TDMV_DCHAN = 24 zaptel.conf: hardhdlc = 24 (or in our case 48 since it's the second span) which means use wanpipe to handle the d-channel hdlc assuming the wanpipe driver has the necessary support (wanpipe on my platform doesn't). > Also, I recommend changing this line: > wbg1 = wanpipe2, , TDM_VOICE, Comment > > To this: > wbg1 = wanpipe2, , TDM_VOICE_API, Comment The sangoma voice API interface isn't available on my platform and shouldn't be necessary when using zaptel. > assuming that this is what you want then you will need to use > ozmod_libpri because the default OpenZAP PRI stack does not > currently support being the network side. Are you sure? Openzap appears to contain implementations for both NT and TE. The configuration file supports specifying either user or network for the mode. Is the NT support currently nonfunctional? I had tried configuring the Cisco as the NT with similar results. > I don't see where timing is specified It's the same T1 which was being used for RBS between FreeSWITCH and the Cisco so that timing (etc) should be okay. No errors are showing up at the physical level and the Cisco reports Layer 1 as active. The trace on the Cisco seems to show Layer 2 coming up (timestamps 22:53:44.264 through 22:54:21.760), then there's a long pause during which no Receive Ready frames are received from FreeSWITCH. At this point the Cisco gets unhappy and marks Layer 2 as down. If nothing obvious comes to anyone's mind, then I'll simply need to trace through the FreeSWITCH ISDN code and see what's going on. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From mszlazak at aol.com Wed Apr 8 16:15:35 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 08 Apr 2009 19:15:35 -0400 Subject: [Freeswitch-users] Problem with originate in javascript. Message-ID: <8CB86B70E569454-8D4-125F@MBLK-M41.sysops.aol.com> I want to run a script with a scheduler but I'm having a problem with how to set up the originate in Javascript. The originate would go something like: originate {id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/12223334444 at 10.0.0.5:5061 GINO_ANS I can get this to work: session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061"); But I want to "drop" that into an extension that runs another script and can't get either of these to work: session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061 GINO_ANS"); session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); Also, will I have problems running the second script from the first script? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/b5ff2c31/attachment.html From wiltingtree at gmail.com Wed Apr 8 18:59:46 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Wed, 8 Apr 2009 21:59:46 -0400 Subject: [Freeswitch-users] Two or more simultaneous calls not Message-ID: Traun, thanks again for your help. I followed your advice and I made some progress! I tested with the latest trunk version and also with 1.0.2, and both exhibited the same behavior. I then tried writing a test script in Lua, and it worked fine. So this meant the problem was in the Python module (I was sure it was some FS config issue). So I started playing with a small test Python script, and I narrowed the problem down to when I'm using the "read" function. Here is my test script: from freeswitch import * def handler(session, args): #answer the call session.answer(); #play a file session.streamFile("long_prompt.mp3") # Test 1 - FAILED! digits = session.read(5, 10, "long_prompt.mp3", 3000, "#") # Test 2 - WORKED OK! #session.getDigits(1,"#",7000) #session.streamFile("long_prompt.mp3") # TEST 3 - WORKED OK! #digits = session.playAndGetDigits(5, 10, 1, 60, "#","long_prompt.mp3", "", "") #hangup session.hangup(); When I uncomment the code under test 1 and I make two simultaneous calls, the initial prompt plays for both calls just fine. But then the second prompt only plays on one of the channels and the other one just has dead air. When the first channel finishes playing the prompt, then the second channel starts playing it. Then I re-comment test 1 and uncomment either test 2 or test 3, Both prompts play just fine for both channels. So I think there may be a bug in the read() function somewhere. I took a look at it, but it's way over my head. Thanks again, Adam >Message: 8 >Date: Thu, 9 Apr 2009 02:36:08 +0430 >From: Traun Leyden >Subject: Re: [Freeswitch-users] Two or more simultaneous calls not >To: freeswitch-users at lists.freeswitch.org >Message-ID: > >Content-Type: text/plain; charset="iso-8859-1" >Hi Adam, >I'm stumped .. I guess you could try the following: >* Try with the trunk version of freeswitch. I don't think it will matter, >but just in case >* Try to simulate the same test with a Lua script. Do you see the same >problem? >If those don't turn up anything, then the next logical step would be >to start adding printf() statements in the mod_python code and >find out where it is getting stuck. In particular around the parts where >it swaps the threadstate in and out. I might be able to create a patch >for you, but try those other tests first. >HTH, >Traun > >> >> Thanks for the response Traun. The version of Python is 2.4.3, and I >> didn't >> build it myself, I installed it with yum. >> The version of Red Hat is 4.1.2-41. >> "import threading" works fine, so I don't think it's a Python threading >> issue. >> The FreeSWITCH version I installed is the >> freeswitch-1.0.3.tar.gz< >> http://files.freeswitch.org/freeswitch-1.0.3.tar.gz> >> located >> at files.freeswitch.org. >> I didn't make any major changes to the configuration; I enabled Python and >> set-up the SIP profile, directory and dialplan. No other changes. >> Any other help would be appreciated, since I really don't know where to >> look. >> >> Thanks, >> Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/aa0f15c8/attachment.html From zhaoxxqq at 163.com Wed Apr 8 19:02:05 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Thu, 9 Apr 2009 10:02:05 +0800 Subject: [Freeswitch-users] Polycom register problem in private address Message-ID: <200904091002045745669@163.com> hi, I use FS server at public Address. I use polycom's IP550 at private address(192.168.0.120), Now there is a problem that the IP550 can not register to FS. But when I use account to eyebeam, the registering is OK. the attachment is my IP 550's config file, I think it must be NAT problem. Can anyone can help me solve it? 2009-04-09 zhaoxxqq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/c0b231a2/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: sip.cfg Type: application/octet-stream Size: 183561 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/c0b231a2/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: phone[0004f2166b56].cfg Type: application/octet-stream Size: 12396 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/c0b231a2/attachment-0003.obj From brian at freeswitch.org Wed Apr 8 20:14:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Apr 2009 22:14:50 -0500 Subject: [Freeswitch-users] Polycom register problem in private address In-Reply-To: <200904091002045745669@163.com> References: <200904091002045745669@163.com> Message-ID: <819E811F-6CC6-407E-B689-2286926CD31D@freeswitch.org> This is because the Polycom doesn't support STUN, RPORT or any other nat traversal technology. You have a couple of choices please review http://wiki.freeswitch.org/wiki/NAT_Traversal Also review the NDLB-force-rport option for the sofia profile to assume rport. CAUTION this breaks things like cisco phones. /b On Apr 8, 2009, at 9:02 PM, zhaoxxqq wrote: > hi, > I use FS server at public Address. I use polycom's IP550 at private > address(192.168.0.120), Now there is a problem that the IP550 can > not register to FS. But when I use account to eyebeam, the > registering is OK. the attachment is my IP 550's config file, I > think it must be NAT problem. Can anyone can help me solve it? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/fec7b13b/attachment.html From solko at gcdf.pl Thu Apr 9 00:51:03 2009 From: solko at gcdf.pl (Szymon Olko) Date: Thu, 09 Apr 2009 09:51:03 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB868AC2870B10-8D4-75@MBLK-M41.sysops.aol.com> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl><8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> <8CB868AC2870B10-8D4-75@MBLK-M41.sysops.aol.com> Message-ID: <49DDA8E7.9060505@gcdf.pl> mszlazak at aol.com pisze: > Na jaki? czas, b?d? uczy? si?, jak radzi? sobie z Tortoise SVN b??d?w. > > M?j polski nie jest zbyt dobre, ale dzi?kuj?. Google pomaga. > Your polish is much better then my english. I think on that list we should stay with english. Where are you from? > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, 8 Apr 2009 10:36 am > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > mszlazak at aol.com pisze: > >> OK , you're SVN updating on a Linux system but I'm using Windows. The > >> very few times I tried with Tortoise SVN I ran into problems were it > >> would fail because of some path not being present or some strange symbol > >> in a file or something else. Since I'm not experienced enough and don't > >> always have the time, I gave up on this approach and just start over > >> again in a different folder then reconfigure the updated FS and transfer > >> files from an older FS. Yup, it sucks. > >> > > > > Yes I'm linux user. > > If you have problems with svn update then you can do your way, make fresh > > checkout every time. After CO copy modules.conf and > > build new version, just copy old config files > to installation directory if it is > > always different one. > > > > That's why I hate gui tools for things like full svn update. > >> > >> > >> -----Original Message----- > >> From: Szymon Olko > > >> To: freeswitch-users at lists.freeswitch.org > >> Sent: Wed, 8 Apr 2009 12:59 am > >> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> 127.0.0.1 and memory consumption > >> > >> mszlazak at aol.com > pisze: > >> > >>> I'm not quite sure what your asking. > >> > >>> Are you saying that I could run the latest FS svn but in a way that uses > >> > >>> my "older" configuration files? If so then I don't, and don't know how > >> > >>> ... blush blush. > >> > >>> If that's the easiest thing to do then please tell me how. > >> > >>> Thanks. Mark. > >> > >> > >> > >> Exactly, I do it that way. > >> > >> For first time I gave installation prefix when configuring FS. You can stay > > with > >> > >> /usr/local/freeswich/. > >> > >> Now every time i call 'make current' and it does not overwrite my > > configuration > >> > >> file. > >> > >> > >> > >> In case of huge changes in modules I copy/merge my config file with the one > > from > >>=2 > 0 > >> svn. I did not have problems with it, because > >> > >> developers makes good default values for new configuration options. > >> > >> > >> > >> I don't know which modules do you use, but in ones I use configuration is not > >> > >> changes a lot, there are new options added which > >> > >> does not break old one. > >> > >> > >> > >> make install do not copy configuration files for me if they are already > >> > >> installed, I have that on production server and all test > >> > >> servers. I assume this is correct behavior and I'm not the only one work like > >> > >> that. > >> > >> I looked in Makefile and it tests for config file before installing, so it > > does > >> > >> not overwrite them. > >> > >> > >> > >> Regards > >> > >> Szymon > >> > >>> > >> > >>> -----Original Message----- > >> > >>> From: Szymon Olko >> > >> > >>> To: freeswitch-users at lists.freeswitch.org > > >> > >>> Sent: Tue, 7 Apr 2009 10:54 pm > >> > >>> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> > >>> 127.0.0.1 and memory consumption > >> > >>>20 > >> > >>> mszlazak at aol.com > > > ?>> pisze: > >> > >>> > >> > >>>> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > >> > >>> > >> > >>>> new trunk and I have to go through vars.xml, etc changing > >> > >>> > >> > >>>> $${local_ip_v4} like you did. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Is there a way to change $${local_ip_v4} in one place. That way one > >> > >>> > >> > >>>> wouldn't have remember all the locations that it needs to be changed? > >> > >>> > >> > >>>> > >> > >>> > >> > >>> My configuration is not updated when I compile new version and install it. Do > > > >> > >>> > >> > >>> you run FS with configuration path pointed to svn > >> > >>> > >> > >>> work dir? > >> > >>> > >> > >>> > >> > >>> > >> > >>>> -----Original Message----- > >> > >>> > >> > >>>> From: Peter P GMX > > > ?>>> > >> > >>> > >> > >>>> To: freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>>> Sent: Tue, 7 Apr 2009 3:52 pm > >> > >>> > >> > >>>> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> > >>> > >> > >>>> 127.0.0.1 and memory consumption > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Thanks Brian, > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> what I was actually looking for was to use a standard SIP soft phone > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> with some additional features. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> I finally manged to make FS li > sten on 127.0.0.1 the following way: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> vars.xml > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> internal.xml > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> The rest is standard configuration. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Now communication Laptop-internal is UDP on port 5060 and external via > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> TLS on p > ort 5081, so I have no open port 5060 to the internet. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Best regards > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Peter > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Brian West schrieb: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> 1st Question: Is that possible or is another solution preferrable? > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > ;> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> Just use FreeSWITCH with mod_portaudio. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> 2nd Question: How can I change the amount of memory FS tries to reserve > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> to an absolute minumum (I only have 1 call at a time). Currently it > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> tries to reserve about 360M if I read that right. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> Thats virtual. Look at RES. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> Best regards > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> Peter > >> > >>> > >> > >>>> > >> > >>> > >> > >>> >>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> Brian West > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> brian at freeswitch.org > > > ?>> > > ?> > >> > >> ?>?>> > > ?> > >> > >> ?>?> > >> > >>> > >> > >>> ?> > > ?>?>?>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > ; > >>>>> -- Meet us a ClueCon! http://www.cluecon.com > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> ------------------------------------------------------------------------ > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> _______________________________________________ > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> Freeswitch-users mailing list > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> Freeswitch-users at lists.freeswitch.org > > > ?>> > > > >> > >> ?> > > ?>?>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> http://www.freeswitch.org > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> _______________________________________________ > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Freeswitch-users mailing list > >> > >>> > >> > >>>> > >> > >>> > >> > >>> >> Freeswitch-users at lists.freeswitch.org > > > ?>> > > > >> > >> ?> > > ?>?>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> http://www.freeswitch.org > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> --------------------- > --------------------------------------------------- > >> > >>> > >> > >>>> *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > >> > >>> > >> > >>>> * > > > > > >> > >> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> ------------------------------------------------------------------------ > >> > >>> > >> > >>>> > >> > >>> > > & > gt; > >>>> _______________________________________________ > >> > >>> > >> > >>>> Freeswitch-users mailing list > >> > >>> > >> > >>>> Freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>>> http://www.freeswitch.org > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> _______________________________________________ > >> > >>> > >> > >>> Freeswitch-users mailing list > >> > >>> > >> > >>> Freeswitch-users at lists.freeswitch.org > > > ers at lists.freeswitch.org ?>> > >> > >>> > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>> http://www.freeswitch.org > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> New Deals on Dell Netbooks - Now starting at $299 > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> > >> > >>> _______________________________________________ > >> > >>> Freeswitch-users mailing list > >> > >>> Freeswitch-users at lists.freeswitch.org > > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> _______________________________________________ > >> > >> Freeswitch-users mailing list > >> > >> Freeswitch-users at lists.freeswitch.org > > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> ------------------------------------------------------------------------ > >> New Deals on Dell Netbooks - Now starting at $299 > >> > >> > >> > >> > >> --------------------------------------- > --------------------------------- > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 bmsword at gmail.com Thu Apr 9 01:13:10 2009 From: bmsword at gmail.com (bmsword) Date: Thu, 9 Apr 2009 16:13:10 +0800 Subject: [Freeswitch-users] Freeswitch as a media server Message-ID: <200904091613018903127@gmail.com> hi,all Can freeswitch be integrated with another softswitch as a media server? if it can, how to configure? thanks! andy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/f9f64ee4/attachment-0001.html From mchlmll at gmail.com Thu Apr 9 01:18:34 2009 From: mchlmll at gmail.com (Michele M) Date: Thu, 9 Apr 2009 10:18:34 +0200 Subject: [Freeswitch-users] How to design my project ? In-Reply-To: <7b197bef0904080956h190a0bb1v6fda1c6965931390@mail.gmail.com> References: <7b197bef0904080956h190a0bb1v6fda1c6965931390@mail.gmail.com> Message-ID: Ciao Giovanni, thanks for the quick answer, but still I can't get if it is possible to use the libfreeswitch w/o running FS to accomplish what I meant. It would need alot of time of development to make my application run as a FS application.Much better would be using libfreeswitch inside my application.But still the question arises:" Is Libfreeswitch enough for having sip endpoints and ASR/TTS API?" Do you have some examples for it? Thanks again Michele 2009/4/8 Giovanni Maruzzelli > Ciao Michele, > > as a start is definitely better (and more gratifying) that you runs > FreeSWITCH. > > Then, if (and only if) there is a compelling reason that justify the > amount of time needed to develop a standalone application, go for it. > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Apr 8, 2009 at 12:34 PM, Michele M wrote: > > Hi there, > > > > I'm quite a newbie about freeswitch. I have an application (IVR) that > > needs to have endpoints SIP to register,answer the calls and transfer > them > > to the right phones.(I( have my own SIP server).Moreover it needs also a > > ASR/TTS API' set to communicate with my ASR/TTS engine ( just for > example > > let's assume it is Cepstral). I'd wouldn't want to have freeswitch > running > > and communicate with it to accomplish that but just to use the > libfreeswitch > > library embedded. As I don't know that much about freeswitch can it be > done? > > or just I need to have freeswitch running as a must? Can somebody point > me > > to the right place where to find example of using library embedded (best > > examples for what I'm trying to do) as I have not found that many? > > > > Thanks in advance > > > > Miki > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/3a710d7f/attachment.html From saigop at gmail.com Thu Apr 9 03:03:40 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Thu, 9 Apr 2009 15:33:40 +0530 Subject: [Freeswitch-users] Freeswitch as a media server In-Reply-To: <200904091613018903127@gmail.com> References: <200904091613018903127@gmail.com> Message-ID: <2ea4d47e0904090303q5006fb37le2b2fc97419b4c41@mail.gmail.com> Yes, you can connect freeswitch with another media gateway like audiocode or any softswitch. you can find here to connect with audiocode http://wiki.freeswitch.org/index.php?title=Configuring_AudioCodes_MP-114/118&printable=yes this is for analog audiocode, you can also connect with same setting with digital audiocodes. On Thu, Apr 9, 2009 at 1:43 PM, bmsword wrote: > hi,all > Can freeswitch be integrated with another softswitch as a media server? > if it can, how to configure? > thanks! > andy. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/479285f2/attachment.html From ceino.no at gmail.com Thu Apr 9 03:28:09 2009 From: ceino.no at gmail.com (Ceino) Date: Thu, 09 Apr 2009 12:28:09 +0200 Subject: [Freeswitch-users] How to design my project ? In-Reply-To: References: <7b197bef0904080956h190a0bb1v6fda1c6965931390@mail.gmail.com> Message-ID: <49DDCDB9.1080509@gmail.com> Hi Miki, I'm not an expert on freeswitch but I'm sure that libfreeswitch can be embedded into c-applications (http://wiki.freeswitch.org/wiki/Embedding_FreeSWITCH). In addition it possible to write part of an application as an loadable module (applications in freeswitch are loadable modules). The reason for this is that a loadable module can use the dialplan framework (for example an incomping call can be routed to your application). It is also a good idea to look into how the scripting languages are used in FS (http://wiki.freeswitch.org/wiki/Languages_for_Call_Control). //Accordingly to voipinfo.org (http://www.voipinfo.org/wiki/view/FreeSwitch) is FS a library which ships with a small executable that loads the library, launches the core, and performs the various tasks that are defined by the modules (freeswitch applications). libfreeswitch includes sofia sip endpoint and ASR/TSS (native or through MRCP). You need to modify some configuration files to enable ASR/TSS (sip endpoint sofia is default). Best Regards Lars Sivertsen Michele M wrote: > Ciao Giovanni, > > thanks for the quick answer, but still I can't get if it is possible > to use the libfreeswitch w/o running FS to accomplish what I meant. > It would need alot of time of development to make my application run > as a FS application.Much better would be using libfreeswitch inside my > application.But still the question arises:" Is Libfreeswitch enough > for having sip endpoints and ASR/TTS API?" > Do you have some examples for it? > > Thanks again > > Michele > > 2009/4/8 Giovanni Maruzzelli > > > Ciao Michele, > > as a start is definitely better (and more gratifying) that you > runs FreeSWITCH. > > Then, if (and only if) there is a compelling reason that justify the > amount of time needed to develop a standalone application, go for it. > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Apr 8, 2009 at 12:34 PM, Michele M > wrote: > > Hi there, > > > > I'm quite a newbie about freeswitch. I have an application > (IVR) that > > needs to have endpoints SIP to register,answer the calls and > transfer them > > to the right phones.(I( have my own SIP server).Moreover it > needs also a > > ASR/TTS API' set to communicate with my ASR/TTS engine ( just > for example > > let's assume it is Cepstral). I'd wouldn't want to have > freeswitch running > > and communicate with it to accomplish that but just to use the > libfreeswitch > > library embedded. As I don't know that much about freeswitch can > it be done? > > or just I need to have freeswitch running as a must? Can > somebody point me > > to the right place where to find example of using library > embedded (best > > examples for what I'm trying to do) as I have not found that many? > > > > Thanks in advance > > > > Miki > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 9 06:09:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Apr 2009 08:09:27 -0500 Subject: [Freeswitch-users] Problem with originate in javascript. In-Reply-To: <8CB86B70E569454-8D4-125F@MBLK-M41.sysops.aol.com> References: <8CB86B70E569454-8D4-125F@MBLK-M41.sysops.aol.com> Message-ID: <191c3a030904090609n440e7a4n80444bcf7902113a@mail.gmail.com> The 2nd 2 examples you provided are invalid, they depict the usage of the originate api command in the context of the constructor to a JS session. If you want to send the call to another extension you have to create the channel like you did in the first example followed by session.execute("transfer", "GINO_ANS XML default"); at which time it would be wise if you deref the session object because its thread will be running in the new extension. A better way would be to do both in one with a single call to the originate api command apiExecute("originate", "{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ 14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); This never gives you a session object it just creates a channel and transfers it to the desired extension. A Documentation Re-factorial Engineer may be able to add it to the relevant page on the wiki if it is not already present. On Wed, Apr 8, 2009 at 6:15 PM, wrote: > I want to run a script with a scheduler but I'm having a problem with how > to set up the originate in Javascript. > > The originate would go something like: > > originate > {id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ > 12223334444 at 10.0.0.5:5061 GINO_ANS > > I can get this to work: > > session = new > Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ > 14082031170 at 10.0.0.5:5061"); > > But I want to "drop" that into an extension that runs another script and > can't get either of these to work: > > session = new > Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ > 14082031170 at 10.0.0.5:5061 GINO_ANS"); > > session = new > Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ > 14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); > > Also, will I have problems running the second script from the first script? > > Thanks. > > > > ------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/d2ce6528/attachment.html From anthony.minessale at gmail.com Thu Apr 9 06:15:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Apr 2009 08:15:38 -0500 Subject: [Freeswitch-users] Two or more simultaneous calls not In-Reply-To: References: Message-ID: <191c3a030904090615j2b736b7bp890c903c00a014f6@mail.gmail.com> easier than you think, Just like Traun had suspected, the methods in the C++ wrapper for the read app was missing the begin and end allow threads calls that are only important to python to tell it to suspend the thread state while FS specific code is being executed. 2 line fix in trunk rev 12958 On Wed, Apr 8, 2009 at 8:59 PM, Adam Wilt wrote: > Traun, thanks again for your help. > I followed your advice and I made some progress! > > I tested with the latest trunk version and also with 1.0.2, and both > exhibited the same behavior. > I then tried writing a test script in Lua, and it worked fine. > So this meant the problem was in the Python module (I was sure it was some > FS config issue). > So I started playing with a small test Python script, and I narrowed the > problem down to when I'm using the "read" function. > Here is my test script: > > from freeswitch import * > def handler(session, args): > #answer the call > session.answer(); > #play a file > session.streamFile("long_prompt.mp3") > # Test 1 - FAILED! > digits = session.read(5, 10, "long_prompt.mp3", 3000, "#") > # Test 2 - WORKED OK! > #session.getDigits(1,"#",7000) > #session.streamFile("long_prompt.mp3") > # TEST 3 - WORKED OK! > #digits = session.playAndGetDigits(5, 10, 1, 60, "#","long_prompt.mp3", > "", "") > #hangup > session.hangup(); > When I uncomment the code under test 1 and I make two simultaneous calls, > the initial prompt plays for both calls just fine. But then the second > prompt only plays on one of the channels and the other one just has dead > air. When the first channel finishes playing the prompt, then the second > channel starts playing it. > > Then I re-comment test 1 and uncomment either test 2 or test 3, Both > prompts play just fine for both channels. > > So I think there may be a bug in the read() function somewhere. I took a > look at it, but it's way over my head. > > Thanks again, > Adam > > > > >Message: 8 > >Date: Thu, 9 Apr 2009 02:36:08 +0430 > >From: Traun Leyden > >Subject: Re: [Freeswitch-users] Two or more simultaneous calls not > >To: freeswitch-users at lists.freeswitch.org > >Message-ID: > > > >Content-Type: text/plain; charset="iso-8859-1" > > >Hi Adam, > > >I'm stumped .. I guess you could try the following: > > >* Try with the trunk version of freeswitch. I don't think it will matter, > >but just in case > > >* Try to simulate the same test with a Lua script. Do you see the same > >problem? > > >If those don't turn up anything, then the next logical step would be > >to start adding printf() statements in the mod_python code and > >find out where it is getting stuck. In particular around the parts where > >it swaps the threadstate in and out. I might be able to create a patch > >for you, but try those other tests first. > > >HTH, > >Traun > > > > >> > >> Thanks for the response Traun. The version of Python is 2.4.3, and I > >> didn't > >> build it myself, I installed it with yum. > >> The version of Red Hat is 4.1.2-41. > >> "import threading" works fine, so I don't think it's a Python threading > >> issue. > >> The FreeSWITCH version I installed is the > >> freeswitch-1.0.3.tar.gz< > >> http://files.freeswitch.org/freeswitch-1.0.3.tar.gz> > >> located > >> at files.freeswitch.org. > >> I didn't make any major changes to the configuration; I enabled Python > and > >> set-up the SIP profile, directory and dialplan. No other changes. > >> Any other help would be appreciated, since I really don't know where to > >> look. > >> > >> Thanks, > >> 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/014c661b/attachment-0001.html From peter.olsson at visionutveckling.se Thu Apr 9 06:34:09 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 9 Apr 2009 15:34:09 +0200 Subject: [Freeswitch-users] Crash in mod_opal when hanging up call... Message-ID: Hello everyone. I've been following this project for quite some time now, but I never got the time to test it. But today I finally had a day off from work, so I could sit and play around with it for some time :) Everything wen't really smooth - even though I built everything from scratch on a Windows machine, including mod_opal (linked against the opal library). And with the docs I found I didn't even have to search Google to get it up and running :) So first of all - what a great job, guys - I'm really impressed, and the code seems really stable as well! My setup right now is FreeSWITCH in Windows XP, with both SIP and H323 enabled. And most of the stuff works just fine. However, I think I've found a bug in mod_opal - it sometimes causes FreeSWITCH to crash when hanging up a call. I think that mod_opal is considered to be in beta stage still, so I'm not all that surprised. :) Check the error found in the log below. Does anyone have any ideas? It's pretty easy to reproduce, just dial in to FreeSWITCH using H323 and hang up the call, for me it happens maybe 2 out of 5 times. I'm using latest SVN trunk versions (checked out today), for both FreeSWITCH and for opal/ptlib. If you need further information, or if I should file a jira case, please get back to me, and I'll try to help out as much as possible. 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:657 FSConnection::OnAlerting() Ring-Ready ! 2009-04-09 14:37:05 [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel opal/in:9999 [c7441e16-394c-d843-9ce4-760786dcecbf] 2009-04-09 14:37:05 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Peter Olsson [172.18.96.100]->9999 in context default 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 initialise opal/in:9999write audio codec G.711-uLaw-64k for connection FSMediaStream-Source-G.711-uLaw-64k 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 initialise opal/in:9999read audio codec G.711-uLaw-64k for connection FSMediaStream-Sink-G.711-uLaw-64k 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:795 FSConnection::OnOpenMediaStream() Channel [opal/in:9999] has been answered Assertion failed: (*frame)->codec != ((void *)0), file ..\..\src\switch_core_io.c, line 202 2009-04-09 14:37:09 [NOTICE] mod_opal.cpp:650 FSConnection::OnReleased() Hangup opal/in:9999 [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-09 14:37:09 [INFO] h323pdu.cxx:1005 H225() Read error (0): From peter.olsson at visionutveckling.se Thu Apr 9 06:37:17 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 9 Apr 2009 15:37:17 +0200 Subject: [Freeswitch-users] Status of Sangoma support in Windows? Message-ID: >From what I've found in the docs and lists, the support for Sangoma (PRI) cards is still not avaiable in the Windows port. Is this planned to be implemented in the future, or will it never be included in Win32? Just a curious thought, since I might need to use some PRI stuff in the future... Regards, Peter Olsson From mike at jerris.com Thu Apr 9 07:05:32 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 9 Apr 2009 10:05:32 -0400 Subject: [Freeswitch-users] Status of Sangoma support in Windows? In-Reply-To: References: Message-ID: <243EF9A5-48F0-45CE-BFB6-8CB54040679B@jerris.com> The code that "should" work for this is on a box at Sangmoa under testing right now on linux. It should be committed as soon as the new driver is released (which the new module will require) at which point it will just need build integration completed and proper testing on windows. Mike On Apr 9, 2009, at 9:37 AM, Peter Olsson wrote: >> From what I've found in the docs and lists, the support for Sangoma >> (PRI) cards is still not avaiable in the Windows port. Is this >> planned to be implemented in the future, or will it never be >> included in Win32? Just a curious thought, since I might need to >> use some PRI stuff in the future... > > Regards, > > Peter Olsson From peter.olsson at visionutveckling.se Thu Apr 9 07:31:29 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 9 Apr 2009 16:31:29 +0200 Subject: [Freeswitch-users] Status of Sangoma support in Windows? In-Reply-To: <243EF9A5-48F0-45CE-BFB6-8CB54040679B@jerris.com> Message-ID: That sounds great - we'll just have to hope for the best :) Thanks for your quick response. //Peter On 09-04-09 16.05, "Michael Jerris" wrote: The code that "should" work for this is on a box at Sangmoa under testing right now on linux. It should be committed as soon as the new driver is released (which the new module will require) at which point it will just need build integration completed and proper testing on windows. Mike On Apr 9, 2009, at 9:37 AM, Peter Olsson wrote: >> From what I've found in the docs and lists, the support for Sangoma >> (PRI) cards is still not avaiable in the Windows port. Is this >> planned to be implemented in the future, or will it never be >> included in Win32? Just a curious thought, since I might need to >> use some PRI stuff in the future... > > Regards, > > Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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:49de027e32935992579123! From peder at networkoblivion.com Thu Apr 9 07:59:56 2009 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Thu, 09 Apr 2009 09:59:56 -0500 Subject: [Freeswitch-users] Polycom register problem in private address In-Reply-To: <819E811F-6CC6-407E-B689-2286926CD31D@freeswitch.org> References: <200904091002045745669@163.com> <819E811F-6CC6-407E-B689-2286926CD31D@freeswitch.org> Message-ID: <49DE0D6C.1080503@networkoblivion.com> You might try entering your external NAT IP into the Polycom config. I've found that if you specify the external IP, Polycom's generally work better thru NAT. This is one area where Cisco is superior to Polycom. On Cisco, you just enable NAT and you don't have to specify the external IP. Of course Cisco has a whole mess of their own issues too. Peder Brian West wrote: > This is because the Polycom doesn't support STUN, RPORT or any other nat > traversal technology. You have a couple of choices please > review http://wiki.freeswitch.org/wiki/NAT_Traversal > > Also review the NDLB-force-rport option for the sofia profile to assume > rport. CAUTION this breaks things like cisco phones. > > /b > > On Apr 8, 2009, at 9:02 PM, zhaoxxqq wrote: > >> hi, >> I use FS server at public Address. I use polycom's IP550 at private >> address(192.168.0.120), Now there is a problem that the IP550 can not >> register to FS. But when I use account to eyebeam, the registering is >> OK. the attachment is my IP 550's config file, I think it must be NAT >> problem. Can anyone can help me solve it? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 9 08:38:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Apr 2009 08:38:25 -0700 Subject: [Freeswitch-users] need help getting ISDN talking to Cisco 3845 In-Reply-To: <200904082219.n38MJ0ld006139@jwlab.FEITH.COM> References: <200904082219.n38MJ0ld006139@jwlab.FEITH.COM> Message-ID: <87f2f3b90904090838i40bf4160l1e4f1bc51f8129be@mail.gmail.com> John, Just curious - why are you using zaptel at all? Does it provide something for you that the wanpipe drivers do not? I use Sangoma only with Sangoma cards and I have a lot of success. -MC On Wed, Apr 8, 2009 at 3:19 PM, John Wehle wrote: > > Okay, a few things. First off, the wanpipe2.conf file has a booboo. > > Don't think so. > > > This line is WRONG: > > TDMV_DCHAN = 0 > > Not exactly. My understanding is you can use either: > > wanpipeX.conf: TDMV_DCHAN = 0 > zaptel.conf: dchan = 24 (or in our case 48 since it's the second span) > > which means use zaptel to handle the d-channel hdlc or > > wanpipeX.conf: TDMV_DCHAN = 24 > zaptel.conf: hardhdlc = 24 (or in our case 48 since it's the second span) > > which means use wanpipe to handle the d-channel hdlc assuming the > wanpipe driver has the necessary support (wanpipe on my platform > doesn't). > > > Also, I recommend changing this line: > > wbg1 = wanpipe2, , TDM_VOICE, Comment > > > > To this: > > wbg1 = wanpipe2, , TDM_VOICE_API, Comment > > The sangoma voice API interface isn't available on my platform > and shouldn't be necessary when using zaptel. > > > assuming that this is what you want then you will need to use > > ozmod_libpri because the default OpenZAP PRI stack does not > > currently support being the network side. > > Are you sure? Openzap appears to contain implementations for > both NT and TE. The configuration file supports specifying > either user or network for the mode. Is the NT support > currently nonfunctional? > > I had tried configuring the Cisco as the NT with similar > results. > > > I don't see where timing is specified > > It's the same T1 which was being used for RBS between > FreeSWITCH and the Cisco so that timing (etc) should > be okay. No errors are showing up at the physical > level and the Cisco reports Layer 1 as active. > > The trace on the Cisco seems to show Layer 2 coming up > (timestamps 22:53:44.264 through 22:54:21.760), then > there's a long pause during which no Receive Ready > frames are received from FreeSWITCH. At this point > the Cisco gets unhappy and marks Layer 2 as down. > > If nothing obvious comes to anyone's mind, then I'll > simply need to trace through the FreeSWITCH ISDN code > and see what's going on. > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | > | John Wehle | Fax: 1-215-540-5495 | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/be59339c/attachment.html From chris at fowler.cc Thu Apr 9 08:51:27 2009 From: chris at fowler.cc (Chris Fowler) Date: Thu, 09 Apr 2009 08:51:27 -0700 Subject: [Freeswitch-users] Polycom register problem in private address Message-ID: <1239292287.27625.1309790615@webmail.messagingengine.com> I'm running FS on Amazons' EC2 compute cloud (AWS) and have 30 Polycom phones working happily in this config. I modified the Internal profile in /usr/local/freeswitch/conf/sip_profiles/internal.xml to include: The phones connect on port 5060 - nothing specical to config in the -phone.cfg file for the phone; just host, port, user/pass. Chris. From brian at freeswitch.org Thu Apr 9 09:09:48 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Apr 2009 11:09:48 -0500 Subject: [Freeswitch-users] Polycom register problem in private address In-Reply-To: <1239292287.27625.1309790615@webmail.messagingengine.com> References: <1239292287.27625.1309790615@webmail.messagingengine.com> Message-ID: <45D90D9C-39D8-4B09-BFE2-77523151F13A@freeswitch.org> Did you request public IP's for your EC2 instance? /b On Apr 9, 2009, at 10:51 AM, Chris Fowler wrote: > I'm running FS on Amazons' EC2 compute cloud (AWS) and have 30 Polycom > phones working happily in this config. > > I modified the Internal profile in > /usr/local/freeswitch/conf/sip_profiles/internal.xml to include: > > > > > The phones connect on port 5060 - nothing specical to config in the > -phone.cfg file for the phone; just host, port, user/pass. > > Chris. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/35cd52f3/attachment-0001.html From msc at freeswitch.org Thu Apr 9 09:16:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Apr 2009 09:16:36 -0700 Subject: [Freeswitch-users] Polycom register problem in private address In-Reply-To: <1239292287.27625.1309790615@webmail.messagingengine.com> References: <1239292287.27625.1309790615@webmail.messagingengine.com> Message-ID: <87f2f3b90904090916g9926c5aj304035602e8c412b@mail.gmail.com> Hey, this would be great info to put on the wiki... (hint hint wink wink nudge nudge) :) -MC On Thu, Apr 9, 2009 at 8:51 AM, Chris Fowler wrote: > I'm running FS on Amazons' EC2 compute cloud (AWS) and have 30 Polycom > phones working happily in this config. > > I modified the Internal profile in > /usr/local/freeswitch/conf/sip_profiles/internal.xml to include: > > > > > The phones connect on port 5060 - nothing specical to config in the > -phone.cfg file for the phone; just host, port, user/pass. > > Chris. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/3e0bb54f/attachment.html From mszlazak at aol.com Thu Apr 9 09:22:49 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Thu, 09 Apr 2009 12:22:49 -0400 Subject: [Freeswitch-users] Problem with originate in javascript. In-Reply-To: <191c3a030904090609n440e7a4n80444bcf7902113a@mail.gmail.com> References: <8CB86B70E569454-8D4-125F@MBLK-M41.sysops.aol.com> <191c3a030904090609n440e7a4n80444bcf7902113a@mail.gmail.com> Message-ID: <8CB87468E0848B5-874-626@WEBMAIL-MY06.sysops.aol.com> Hi Tony, But I thought we settled on "Janitor." ;-) BTW, it was the other point about keeping the FS founders involved or not in the documentation process that concerned much much more. That was the big issue that got me going on that thread. Anyway, I appreciate your help and will do some "document engineering" but need one further elaboration on de-referencing the original session object since I tried the execute("transfer" ...) before and couldn't get that to work. Can you show me an example and I can then put up both approaches. Mark. -----Original Message----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thu, 9 Apr 2009 6:09 am Subject: Re: [Freeswitch-users] Problem with originate in javascript. The 2nd 2 examples you provided are invalid, they depict the usage of the originate api command in the context of the constructor to a JS session. If you want to send the call to another extension you have to create the channel like you did in the first example followed by session.execute("transfer", "GINO_ANS XML default"); at which time it would be wise if you deref the session object because its thread will be running in the new extension. A better way would be to do both in one with a single call to the originate api command apiExecute("originate", "{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); This never gives you a session object it just creates a channel and transfers it to the desired extension. A Documentation Re-factorial Engineer may be able to add it to the relevant page on the wiki if it is not already present. On Wed, Apr 8, 2009 at 6:15 PM, wrote: I want to run a script with a scheduler but I'm having a problem with how to set up the originate in Javascript. The originate would go something like: originate {id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/12223334444 at 10.0.0.5:5061 GINO_ANS I can get this to work: session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061"); But I want to "drop" that into an extension that runs another script and can't get either of these to work: session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061 GINO_ANS"); session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); Also, will I have problems running the second script from the first script? Thanks. New Deals on Dell Netbooks - Now starting at $299 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/3a38fa31/attachment.html From chris at fowler.cc Thu Apr 9 09:30:18 2009 From: chris at fowler.cc (Chris Fowler) Date: Thu, 09 Apr 2009 09:30:18 -0700 Subject: [Freeswitch-users] Polycom register problem in private address In-Reply-To: <45D90D9C-39D8-4B09-BFE2-77523151F13A@freeswitch.org> References: <1239292287.27625.1309790615@webmail.messagingengine.com> <45D90D9C-39D8-4B09-BFE2-77523151F13A@freeswitch.org> Message-ID: <1239294618.4932.1309797565@webmail.messagingengine.com> Brian: Did you request public IP's for your EC2 instance? Yes; there is an Elastic IP (EIP) associated with the instance. Also specify the EIP in vars.xml >> Re: Wiki Yup I need to get on this. FWIW - I work for RightScale; our computer room is empty except for routers and switches. *Everything* else lives in the Cloud :-) Cheers, Chris. From anthony.minessale at gmail.com Thu Apr 9 09:40:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Apr 2009 11:40:32 -0500 Subject: [Freeswitch-users] Problem with originate in javascript. In-Reply-To: <8CB87468E0848B5-874-626@WEBMAIL-MY06.sysops.aol.com> References: <8CB86B70E569454-8D4-125F@MBLK-M41.sysops.aol.com> <191c3a030904090609n440e7a4n80444bcf7902113a@mail.gmail.com> <8CB87468E0848B5-874-626@WEBMAIL-MY06.sysops.aol.com> Message-ID: <191c3a030904090940l6e6ad6c7wfb64cf6fd48afd30@mail.gmail.com> I think I forgot to mention you need to session.setAutoHangup(false) to stop the channel from being auto-hungup when it goes out of scope of the script. session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ 14082031170 at 10.0.0.5:5061"); session.setAutoHangup(false); session.execute("transfer", "GINO_ANS XML default"); session = undefined; On Thu, Apr 9, 2009 at 11:22 AM, wrote: > Hi Tony, > > But I thought we settled on "Janitor." ;-) > BTW, it was the other point about keeping the FS founders involved or not > in the documentation process that concerned much much more. That was the big > issue that got me going on that thread. > > Anyway, I appreciate your help and will do some "document engineering" but > need one further elaboration on de-referencing the original session object > since I tried the execute("transfer" ...) before and couldn't get that to > work. Can you show me an example and I can then put up both approaches. > > Mark. > > > -----Original Message----- > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, 9 Apr 2009 6:09 am > Subject: Re: [Freeswitch-users] Problem with originate in javascript. > > The 2nd 2 examples you provided are invalid, they depict the usage of the > originate api command in the context of the constructor > to a JS session. > > If you want to send the call to another extension you have to create the > channel like you did in the first example followed by > session.execute("transfer", "GINO_ANS XML default"); > at which time it would be wise if you deref the session object because its > thread will be running in the new extension. > > > A better way would be to do both in one with a single call to the originate > api command > > apiExecute("originate", > "{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ > 14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); > > This never gives you a session object it just creates a channel and > transfers it to the desired extension. > > > A Documentation Re-factorial Engineer may be able to add it to the relevant > page on the wiki if it is not already present. > > > > On Wed, Apr 8, 2009 at 6:15 PM, wrote: > >> I want to run a script with a scheduler but I'm having a problem with how >> to set up the originate in Javascript. >> >> The originate would go something like: >> >> originate >> {id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ >> 12223334444 at 10.0.0.5:5061 GINO_ANS >> >> I can get this to work: >> >> session = new >> Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ >> 14082031170 at 10.0.0.5:5061"); >> >> But I want to "drop" that into an extension that runs another script and >> can't get either of these to work: >> >> session = new >> Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ >> 14082031170 at 10.0.0.5:5061 GINO_ANS"); >> >> session = new >> Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ >> 14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); >> >> Also, will I have problems running the second script from the first >> script? >> >> Thanks. >> >> >> >> ------------------------------ >> New Deals on Dell Netbooks - Now starting at $299 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/82ead3fe/attachment-0001.html From anthony.minessale at gmail.com Thu Apr 9 15:03:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Apr 2009 17:03:05 -0500 Subject: [Freeswitch-users] Crash in mod_opal when hanging up call... In-Reply-To: References: Message-ID: <191c3a030904091503g69885e9aj191da55d1162dc62@mail.gmail.com> can you try the latest trunk again r12975 if it's still a problem please open a jira on the issue http://jira.freeswitch.org On Thu, Apr 9, 2009 at 8:34 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Hello everyone. I've been following this project for quite some time now, > but I never got the time to test it. But today I finally had a day off from > work, so I could sit and play around with it for some time :) Everything > wen't really smooth - even though I built everything from scratch on a > Windows machine, including mod_opal (linked against the opal library). And > with the docs I found I didn't even have to search Google to get it up and > running :) So first of all - what a great job, guys - I'm really impressed, > and the code seems really stable as well! > > My setup right now is FreeSWITCH in Windows XP, with both SIP and H323 > enabled. And most of the stuff works just fine. However, I think I've found > a bug in mod_opal - it sometimes causes FreeSWITCH to crash when hanging up > a call. I think that mod_opal is considered to be in beta stage still, so > I'm not all that surprised. :) Check the error found in the log below. Does > anyone have any ideas? It's pretty easy to reproduce, just dial in to > FreeSWITCH using H323 and hang up the call, for me it happens maybe 2 out of > 5 times. > > I'm using latest SVN trunk versions (checked out today), for both > FreeSWITCH and for opal/ptlib. > > If you need further information, or if I should file a jira case, please > get back to me, and I'll try to help out as much as possible. > > 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:657 FSConnection::OnAlerting() > Ring-Ready ! > 2009-04-09 14:37:05 [NOTICE] switch_channel.c:597 switch_channel_set_name() > New Channel opal/in:9999 [c7441e16-394c-d843-9ce4-760786dcecbf] > 2009-04-09 14:37:05 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing Peter Olsson [172.18.96.100]->9999 in context default > 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 > initialise opal/in:9999write audio codec G.711-uLaw-64k for connection > FSMediaStream-Source-G.711-uLaw-64k > 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 > initialise opal/in:9999read audio codec G.711-uLaw-64k for connection > FSMediaStream-Sink-G.711-uLaw-64k > 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:795 > FSConnection::OnOpenMediaStream() Channel [opal/in:9999] has been answered > Assertion failed: (*frame)->codec != ((void *)0), file > ..\..\src\switch_core_io.c, line 202 > 2009-04-09 14:37:09 [NOTICE] mod_opal.cpp:650 FSConnection::OnReleased() > Hangup opal/in:9999 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-04-09 14:37:09 [INFO] h323pdu.cxx:1005 H225() Read error (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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/ce9178e8/attachment.html From tleyden at branchcut.com Thu Apr 9 22:35:51 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Fri, 10 Apr 2009 10:05:51 +0430 Subject: [Freeswitch-users] Two or more simultaneous calls not Message-ID: Hey you beat me to it. I was going to have a look this morning but had no internet because some asswipe cut a bunch of fiber optic cables and took out phone/internet for a big part of the bay area. I haven't tried your patch yet, but I see something that looks suspect: http://fisheye.freeswitch.org/browse/FreeSWITCH/src/switch_cpp.cpp?r=12958 In playAndGetDigits() there are now two calls to begin_allow_threads() (line 778 and 780) followed by only one call to end_allow_threads() (line 793) Also I guess it would have better to test against JS, since it should have had the same bug right? Lua just ignores the threadswapping stuff but IIRC javascript uses it in much the same way as python. Or did I miss something? > Message: 5 > Date: Thu, 9 Apr 2009 08:15:38 -0500 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Two or more simultaneous calls not > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030904090615j2b736b7bp890c903c00a014f6 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > easier than you think, > > Just like Traun had suspected, the methods in the C++ wrapper for the read > app was missing the begin and end allow threads calls > that are only important to python to tell it to suspend the thread state > while FS specific code is being executed. > > 2 line fix in trunk rev 12958 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090410/1dfcf048/attachment.html From excelsio at gmx.net Fri Apr 10 00:50:42 2009 From: excelsio at gmx.net (excelsio at gmx.net) Date: Fri, 10 Apr 2009 09:50:42 +0200 Subject: [Freeswitch-users] encryption gateway/proxy with freeswitch? Message-ID: <20090410075042.94340@gmx.net> Hi, the Alcatel OmniPCX Enterprise (OXE) of one of our customers seems to support the following encyption scenarios: - IP-phone on Alcatel OXE <=SRTP=> IP-phone on Alcatel OXE - IP-phone <=SIPS/SRTP=> Alcatel OXE <=> landline phone Unfortunately the Alcatel OXE doesn?t support SIPS/SRTP encryption betwenn itself and a SIP provider. So, for now it looks like: - Alcatel OXE <=SIP/RTP=> SIP provider The goal is, to encrypt that traffic, too. The SIP provider does support SRTP. So I?m asking myself whether to place a freeswitch between both which proxies and also encrypts the traffic? - Alcatel OXE <=SIP/STP> Freeswitch <=(SIPS)/SRTP=> SIP provider Has someone done something like this already? thanks in advance Michael From peter.olsson at visionutveckling.se Fri Apr 10 01:43:48 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 10 Apr 2009 10:43:48 +0200 Subject: [Freeswitch-users] Crash in mod_opal when hanging up call... In-Reply-To: <191c3a030904091503g69885e9aj191da55d1162dc62@mail.gmail.com> Message-ID: Thanks for the reply. This update got rid of the error in the log, but FreeSWITCH still crashes. I've files a jira cace about it (MODOPAL-3). On 09-04-10 00.03, "Anthony Minessale" wrote: can you try the latest trunk again r12975 if it's still a problem please open a jira on the issue http://jira.freeswitch.org On Thu, Apr 9, 2009 at 8:34 AM, Peter Olsson wrote: Hello everyone. I've been following this project for quite some time now, but I never got the time to test it. But today I finally had a day off from work, so I could sit and play around with it for some time :) Everything wen't really smooth - even though I built everything from scratch on a Windows machine, including mod_opal (linked against the opal library). And with the docs I found I didn't even have to search Google to get it up and running :) So first of all - what a great job, guys - I'm really impressed, and the code seems really stable as well! My setup right now is FreeSWITCH in Windows XP, with both SIP and H323 enabled. And most of the stuff works just fine. However, I think I've found a bug in mod_opal - it sometimes causes FreeSWITCH to crash when hanging up a call. I think that mod_opal is considered to be in beta stage still, so I'm not all that surprised. :) Check the error found in the log below. Does anyone have any ideas? It's pretty easy to reproduce, just dial in to FreeSWITCH using H323 and hang up the call, for me it happens maybe 2 out of 5 times. I'm using latest SVN trunk versions (checked out today), for both FreeSWITCH and for opal/ptlib. If you need further information, or if I should file a jira case, please get back to me, and I'll try to help out as much as possible. 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:657 FSConnection::OnAlerting() Ring-Ready ! 2009-04-09 14:37:05 [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel opal/in:9999 [c7441e16-394c-d843-9ce4-760786dcecbf] 2009-04-09 14:37:05 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Peter Olsson [172.18.96.100]->9999 in context default 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 initialise opal/in:9999write audio codec G.711-uLaw-64k for connection FSMediaStream-Source-G.711-uLaw-64k 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 initialise opal/in:9999read audio codec G.711-uLaw-64k for connection FSMediaStream-Sink-G.711-uLaw-64k 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:795 FSConnection::OnOpenMediaStream() Channel [opal/in:9999] has been answered Assertion failed: (*frame)->codec != ((void *)0), file ..\..\src\switch_core_io.c, line 202 2009-04-09 14:37:09 [NOTICE] mod_opal.cpp:650 FSConnection::OnReleased() Hangup opal/in:9999 [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-09 14:37:09 [INFO] h323pdu.cxx:1005 H225() Read error (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 From peter.olsson at visionutveckling.se Fri Apr 10 01:46:54 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 10 Apr 2009 10:46:54 +0200 Subject: [Freeswitch-users] mod_opal and DTMF... Message-ID: When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem to detect DTMF. I'm not sure if this is a setting somewhere in the config files, or if it's a bug. The test scenario is simple - use the default FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't detect any DTMF tones. Regards, Peter Olsson From peter.olsson at visionutveckling.se Fri Apr 10 01:59:25 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 10 Apr 2009 10:59:25 +0200 Subject: [Freeswitch-users] Some spidermonkey modules fails to load in Windows Message-ID: The spidermonkey modules core_db/odbc, curl, socket and teletone fails to load in Windows. They just return error 1271 (Sym Error). I'm not sure if this is a known issue, or if just doesn't work in Windows :) I've been using the latest SVN when trying this. Any ideas anyone? :) Regards, Peter Olsson From peter.olsson at visionutveckling.se Fri Apr 10 03:05:06 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 10 Apr 2009 12:05:06 +0200 Subject: [Freeswitch-users] How to find/build FreeSWITCH.Managed.dll? Message-ID: Hello again! When trying to load the mod_managed module it get an error that it can't find FreeSWITCH.Managed.dll. So my question is simply - where do I find this file, or how do I build it? I'm using the VC++ Express edition when building, so I guess I also have to install the C# edition - will this solve my problem? Sorry for asking stupid questions here - but I've just been playing around with FreeSWITCH for a day or so :) Regards, Peter Olsson From mike at jerris.com Fri Apr 10 05:19:18 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Apr 2009 08:19:18 -0400 Subject: [Freeswitch-users] Two or more simultaneous calls not In-Reply-To: References: Message-ID: <2DBE8205-E99A-4387-A727-B922B44A3236@jerris.com> On Apr 10, 2009, at 1:35 AM, Traun Leyden wrote: > > Hey you beat me to it. I was going to have a look this morning but > had no internet because some asswipe cut a bunch of fiber optic cables > and took out phone/internet for a big part of the bay area. > > I haven't tried your patch yet, but I see something that looks > suspect: > > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/switch_cpp.cpp?r=12958 > > In playAndGetDigits() there are now two calls to begin_allow_threads() > (line 778 and 780) followed by only one call to end_allow_threads() > (line 793) > > Also I guess it would have better to test against JS, since it > should have > had the same bug right? Lua just ignores the threadswapping stuff but > IIRC javascript uses it in much the same way as python. Or did I > miss something? > It's similar, but we don't use swig for javascript and we don't use switch_cpp so it would not have reproduced the issue. MIke -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090410/37d3794b/attachment.html From mike at jerris.com Fri Apr 10 05:26:29 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Apr 2009 08:26:29 -0400 Subject: [Freeswitch-users] Two or more simultaneous calls not In-Reply-To: References: Message-ID: On Apr 10, 2009, at 1:35 AM, Traun Leyden wrote: > > Hey you beat me to it. I was going to have a look this morning but > had no internet because some asswipe cut a bunch of fiber optic cables > and took out phone/internet for a big part of the bay area. > > I haven't tried your patch yet, but I see something that looks > suspect: > > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/switch_cpp.cpp?r=12958 > > In playAndGetDigits() there are now two calls to begin_allow_threads() > (line 778 and 780) followed by only one call to end_allow_threads() > (line 793) Extra line removed, I did notice when looking in that file other methods that probably need begin/end, for example the blocking pop in event consumer. Would you mind going through the rest and seeing if their are other obvious misses? Mike > > Also I guess it would have better to test against JS, since it > should have > had the same bug right? Lua just ignores the threadswapping stuff but > IIRC javascript uses it in much the same way as python. Or did I > miss something? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090410/f276344b/attachment.html From mike at jerris.com Fri Apr 10 05:29:18 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Apr 2009 08:29:18 -0400 Subject: [Freeswitch-users] Some spidermonkey modules fails to load in Windows In-Reply-To: References: Message-ID: <115D04CB-3FD5-4018-8667-F5F45AD405E6@jerris.com> On Apr 10, 2009, at 4:59 AM, Peter Olsson wrote: > The spidermonkey modules core_db/odbc, curl, socket and teletone > fails to load in Windows. They just return error 1271 (Sym Error). > I'm not sure if this is a known issue, or if just doesn't work in > Windows :) I've been using the latest SVN when trying this. > > Any ideas anyone? :) This was fixed in svn yesterday. Mike From mike at jerris.com Fri Apr 10 05:33:14 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Apr 2009 08:33:14 -0400 Subject: [Freeswitch-users] How to find/build FreeSWITCH.Managed.dll? In-Reply-To: References: Message-ID: <04C59CF7-6285-4CCD-818A-B71360F91B8E@jerris.com> On Apr 10, 2009, at 6:05 AM, Peter Olsson wrote: > When trying to load the mod_managed module it get an error that it > can't find FreeSWITCH.Managed.dll. > > So my question is simply - where do I find this file, or how do I > build it? I'm using the VC++ Express edition when building, so I > guess I also have to install the C# edition - will this solve my > problem? > > Sorry for asking stupid questions here - but I've just been playing > around with FreeSWITCH for a day or so :) Unfortunately Express edition does not allow for mixed soulutions so you need to go build the managed dll manually. The file is in src/mod/ languages/mod_managed/managed. Mike From peter.olsson at visionutveckling.se Fri Apr 10 06:11:35 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 10 Apr 2009 15:11:35 +0200 Subject: [Freeswitch-users] Some spidermonkey modules fails to load in Windows In-Reply-To: <115D04CB-3FD5-4018-8667-F5F45AD405E6@jerris.com> Message-ID: I looked into the SVN logs early this morning, and found out that something was changed for this. However, the problem still exists for me, even though I make a clean and full rebuild of FreeSWITCH. Peter On 09-04-10 14.29, "Michael Jerris" wrote: On Apr 10, 2009, at 4:59 AM, Peter Olsson wrote: > The spidermonkey modules core_db/odbc, curl, socket and teletone > fails to load in Windows. They just return error 1271 (Sym Error). > I'm not sure if this is a known issue, or if just doesn't work in > Windows :) I've been using the latest SVN when trying this. > > Any ideas anyone? :) This was fixed in svn yesterday. 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 !DSPAM:49df3d7c32935147211740! From dujinfang at gmail.com Fri Apr 10 11:38:32 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 11 Apr 2009 02:38:32 +0800 Subject: [Freeswitch-users] skypiax Round Robin interface Message-ID: Hi, I made a patch, so skypiax is possible to do a RR hunt besides the sequential interface ANY. Usage: originate skypiax/RR/other_skype_name sk list http://jira.freeswitch.org/browse/MODENDP-211 From mgg at giagnocavo.net Fri Apr 10 12:47:55 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 10 Apr 2009 15:47:55 -0400 Subject: [Freeswitch-users] How to find/build FreeSWITCH.Managed.dll? In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C67025DE8F196@mse17be1.mse17.exchange.ms> Yes you will need to compile the managed one with C#. It should be enough to go to the freeswitch\src\mod\languages\mod_managed\managed directory and execute msbuild. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Friday, April 10, 2009 4:05 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How to find/build FreeSWITCH.Managed.dll? Hello again! When trying to load the mod_managed module it get an error that it can't find FreeSWITCH.Managed.dll. So my question is simply - where do I find this file, or how do I build it? I'm using the VC++ Express edition when building, so I guess I also have to install the C# edition - will this solve my problem? Sorry for asking stupid questions here - but I've just been playing around with FreeSWITCH for a day or so :) Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 Fri Apr 10 21:37:50 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 10 Apr 2009 23:37:50 -0500 Subject: [Freeswitch-users] announcing mod_cidlookup Message-ID: Another itch scratched. :) I just committed a new module mod_cidlookup. mod_cidlookup allows one to: * lookup number->name mapping in a local database * lookup number->name mapping from a URL * cache the results of the URL lookup in memcache The URL lookup is useful when using third party number to name lookup services. Read more about it at: http://wiki.freeswitch.org/wiki/Mod_cidlookup File a jira or email me if you have issues. -- -Rupa From mszlazak at aol.com Sat Apr 11 00:24:04 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 11 Apr 2009 03:24:04 -0400 Subject: [Freeswitch-users] Kill, close or reset a channel that remains with "show channels count" Message-ID: <8CB888DA0788FAD-8D8-3422@mblk-d38.sysops.aol.com> Initially, "show" "channels count" gives "0 total." I then spawn a session from javascript thus: apiExecute("originate", "{id_name='" + call['Caller Name'] + "',id_number=" + call["Caller Number"] + "}sofia/gateway/spa3102PSTN/" + "1" + call["Caller Number"] + "@10.0.0.5:5061 '&javascript(reminder.js \'${id_name}\' ${id_number})'"); Next, I'd like to automatically spawn another session when my channels show "0 total" with apiExecute("show", "channels count"); I've set up a (loop with a msleep) to check when apiExecute("show", "channels count") becomes zero but it never does after the first call and stays at 1. This seems to "mess-up" making the next call. Thanks. Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/e40e01fe/attachment.html From anthony.minessale at gmail.com Sat Apr 11 09:30:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 11 Apr 2009 11:30:08 -0500 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: References: Message-ID: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> see if it works in latest trunk please. On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem > to detect DTMF. I'm not sure if this is a setting somewhere in the config > files, or if it's a bug. The test scenario is simple - use the default > FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't > detect any DTMF tones. > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/c1f7f9ff/attachment-0001.html From ojab at ojab.ru Sat Apr 11 09:49:43 2009 From: ojab at ojab.ru (ojab) Date: Sat, 11 Apr 2009 20:49:43 +0400 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> References: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> Message-ID: mod_opal in latest trunk is broken http://jira.freeswitch.org/browse/MODOPAL-4 so dtmf definitely doen't work ._. //wbr On Sat, Apr 11, 2009 at 8:30 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > see if it works in latest trunk please. > > > > On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > >> When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem >> to detect DTMF. I'm not sure if this is a setting somewhere in the config >> files, or if it's a bug. The test scenario is simple - use the default >> FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't >> detect any DTMF tones. >> >> Regards, >> >> Peter Olsson >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/cf7f74a9/attachment.html From ojab at ojab.ru Sat Apr 11 09:51:48 2009 From: ojab at ojab.ru (ojab) Date: Sat, 11 Apr 2009 20:51:48 +0400 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: References: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> Message-ID: oops, sorry for the noise, fxd now. On Sat, Apr 11, 2009 at 8:49 PM, ojab wrote: > mod_opal in latest trunk is broken > http://jira.freeswitch.org/browse/MODOPAL-4 so dtmf definitely doen't work > ._. > > //wbr > > > On Sat, Apr 11, 2009 at 8:30 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> see if it works in latest trunk please. >> >> >> >> On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson < >> peter.olsson at visionutveckling.se> wrote: >> >>> When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem >>> to detect DTMF. I'm not sure if this is a setting somewhere in the config >>> files, or if it's a bug. The test scenario is simple - use the default >>> FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't >>> detect any DTMF tones. >>> >>> Regards, >>> >>> Peter Olsson >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> 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/ >> >> 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 >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/f6481835/attachment.html From diego.viola at gmail.com Sat Apr 11 16:50:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 11 Apr 2009 19:50:02 -0400 Subject: [Freeswitch-users] Can't send commands with event socket outbound Message-ID: <86a32abc0904111650n2b72c55eyc33665d014286a77@mail.gmail.com> Hello, I'm testing event socket outbound and whenever I use this: -bash-3.2# nc -v -l 127.0.0.1 8084 I send a call to the socket from my dialplan: I can receive just fine but I can't send events from the nc cli. I tried with: connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n But nothing happens, what's wrong? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/4807ed5b/attachment.html From diego.viola at gmail.com Sat Apr 11 16:59:22 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 11 Apr 2009 19:59:22 -0400 Subject: [Freeswitch-users] Can't send commands with event socket outbound In-Reply-To: <86a32abc0904111650n2b72c55eyc33665d014286a77@mail.gmail.com> References: <86a32abc0904111650n2b72c55eyc33665d014286a77@mail.gmail.com> Message-ID: <86a32abc0904111659m797cb28er4731cbc39f1d0803@mail.gmail.com> Never mind, problem solved. On Sat, Apr 11, 2009 at 7:50 PM, Diego Viola wrote: > Hello, > > I'm testing event socket outbound and whenever I use this: > > -bash-3.2# nc -v -l 127.0.0.1 8084 > > I send a call to the socket from my dialplan: > > > > I can receive just fine but I can't send events from the nc cli. > > I tried with: > > connect\n\n > sendmsg > call-command: execute > execute-app-name: answer\n\n > > But nothing happens, what's wrong? > > Thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/abdb5354/attachment.html From diego.viola at gmail.com Sat Apr 11 19:36:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 11 Apr 2009 22:36:47 -0400 Subject: [Freeswitch-users] Question about play_and_get_digits in mod_dptools Message-ID: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> Hi all, I want to use play_and_get_digits from mod_dptools and have some questions about it. I have used playAndGetDigits() in Lua but I see the syntax in mod_dptools is a bit different and I got a bit confused. I see the syntax in the play_and_get_digits from the mod_dptools is something like this: switch_play_and_get_digits(session, min_digits, max_digits, max_tries, timeout, valid_terminators, prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, sizeof(digit_buffer), digits_regex} Can you please explain to me what the session parameter is? And will this allow me to use a phrase macro so I can call my IVR instead of calling a regular file? This is how I use the playAndGetDigits in Lua: digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", "", "\\d+"); I call a phrase macro instead of playing a file, can I do the same with play_and_get_digits from mod_dptools? and please explain me what the session parameter in play_and_get_digits is. Thanks, Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/4318c6bc/attachment.html From diego.viola at gmail.com Sat Apr 11 19:38:41 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 11 Apr 2009 22:38:41 -0400 Subject: [Freeswitch-users] Question about play_and_get_digits in mod_dptools In-Reply-To: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> References: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> Message-ID: <86a32abc0904111938o2e533481x6fdd0318940ca8a@mail.gmail.com> If you give me some examples of how to use play_and_get_digits in mod_dptools I will document it here. http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_play_and_get_digits Diego On Sat, Apr 11, 2009 at 10:36 PM, Diego Viola wrote: > Hi all, > > I want to use play_and_get_digits from mod_dptools and have some questions > about it. > > I have used playAndGetDigits() in Lua but I see the syntax in mod_dptools > is a bit different and I got a bit confused. > > I see the syntax in the play_and_get_digits from the mod_dptools is > something like this: > > switch_play_and_get_digits(session, min_digits, max_digits, > max_tries, timeout, valid_terminators, > > prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, > sizeof(digit_buffer), digits_regex} > > Can you please explain to me what the session parameter is? And will this > allow me to use a phrase macro so I can call my IVR instead of calling a > regular file? > > This is how I use the playAndGetDigits in Lua: > > digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", > "", "\\d+"); > > I call a phrase macro instead of playing a file, can I do the same with > play_and_get_digits from mod_dptools? and please explain me what the session > parameter in play_and_get_digits is. > > Thanks, > > Diego > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/2607554e/attachment-0001.html From diego.viola at gmail.com Sat Apr 11 20:47:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 11 Apr 2009 23:47:02 -0400 Subject: [Freeswitch-users] Question about play_and_get_digits in mod_dptools In-Reply-To: <86a32abc0904111938o2e533481x6fdd0318940ca8a@mail.gmail.com> References: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> <86a32abc0904111938o2e533481x6fdd0318940ca8a@mail.gmail.com> Message-ID: <86a32abc0904112047k2bc0df91ib6b6a2c703f6acd5@mail.gmail.com> Mike just answered. 23:45 < MikeJ> diegoviola: the syntax is documented 23:45 < MikeJ> SWITCH_ADD_APP(app_interface, "play_and_get_digits", "Play and get Digits", "Play and get Digits", 23:45 < MikeJ> IIII play_and_get_digits_function, " ", SAF_NONE); Thanks guys. Diego On Sat, Apr 11, 2009 at 10:38 PM, Diego Viola wrote: > If you give me some examples of how to use play_and_get_digits in > mod_dptools I will document it here. > > > http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_play_and_get_digits > > Diego > > > On Sat, Apr 11, 2009 at 10:36 PM, Diego Viola wrote: > >> Hi all, >> >> I want to use play_and_get_digits from mod_dptools and have some questions >> about it. >> >> I have used playAndGetDigits() in Lua but I see the syntax in mod_dptools >> is a bit different and I got a bit confused. >> >> I see the syntax in the play_and_get_digits from the mod_dptools is >> something like this: >> >> switch_play_and_get_digits(session, min_digits, max_digits, >> max_tries, timeout, valid_terminators, >> >> prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, >> sizeof(digit_buffer), digits_regex} >> >> Can you please explain to me what the session parameter is? And will this >> allow me to use a phrase macro so I can call my IVR instead of calling a >> regular file? >> >> This is how I use the playAndGetDigits in Lua: >> >> digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", >> "", "\\d+"); >> >> I call a phrase macro instead of playing a file, can I do the same with >> play_and_get_digits from mod_dptools? and please explain me what the session >> parameter in play_and_get_digits is. >> >> Thanks, >> >> Diego >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/a1faec3d/attachment.html From pawzlion at gmail.com Sat Apr 11 21:55:14 2009 From: pawzlion at gmail.com (David Robinson) Date: Sun, 12 Apr 2009 14:55:14 +1000 Subject: [Freeswitch-users] Selecting a particular outgoing gateway ? In-Reply-To: References: Message-ID: <6ECDE49F-52F1-48E6-95CA-716A16CF5742@gmail.com> I have configured two providers in sip_profiles/external/example.xml as follows: I have put this in dialplan/defaul.xml to dial out. Regexp matches the first extension if it's a mobile. I want it to route to Pennytelm. voicepulse_pt. Is this how I should be doing this ? I want to specify a different gateway for a different rexep. Please give me some idea what path I should take. David From rupa at rupa.com Sat Apr 11 22:27:27 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Sun, 12 Apr 2009 00:27:27 -0500 Subject: [Freeswitch-users] Selecting a particular outgoing gateway ? In-Reply-To: <6ECDE49F-52F1-48E6-95CA-716A16CF5742@gmail.com> References: <6ECDE49F-52F1-48E6-95CA-716A16CF5742@gmail.com> Message-ID: Sure, you can simply set your regexp in your dialplan. Your example routes numbers starting with 04 to voicepulse_pt 4 or 10 digit numbers go to voicepulse. Now, why you named your gateways voicepulse when neither are using voicepulse, I'll let you ponder. :) If your needs are more complex and/or you have many routes, look into using mod_lcr. I have routes loaded from 2 providers in my tables (teliax and vitelity). phone=> select count(*) from lcr; count ------- 15425 (1 row) Queries against this datbase (postgresql 8.3, using prefix module on a lower end box in the middle of a rsync backup) for lcr info are on the order of 5 milliseconds. http://wiki.freeswitch.org/wiki/Mod_lcr On Sat, Apr 11, 2009 at 11:55 PM, David Robinson wrote: > I have configured two providers in sip_profiles/external/example.xml > as follows: > > > ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? > > ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? > > > I have put this in dialplan/defaul.xml to dial out. Regexp matches the > first extension if it's a mobile. I want it to route to Pennytelm. > voicepulse_pt. > > ? ? > ? ? > ? ? ? data="effective_caller_id_number=11111111111"/> > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? > > ? ? > ? ? > ? ? ? data="effective_caller_id_number=99999999999"/> > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? > > Is this how I should be doing this ? I want to specify a different > gateway for a different rexep. Please give me some idea what path I > should take. > > David > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 From jason at jasonjgw.net Sat Apr 11 22:29:35 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 12 Apr 2009 15:29:35 +1000 Subject: [Freeswitch-users] Selecting a particular outgoing gateway ? In-Reply-To: <6ECDE49F-52F1-48E6-95CA-716A16CF5742@gmail.com> References: <6ECDE49F-52F1-48E6-95CA-716A16CF5742@gmail.com> Message-ID: <20090412052935.GA7458@jdc.jasonjgw.net> David Robinson wrote: > Is this how I should be doing this ? I want to specify a different > gateway for a different rexep. Please give me some idea what path I > should take. Make sure that FreeSWITCH actually reaches your extensions while searching the dial plan. Order is important: if another extension is matched first, and continue="true" is not specified in that extension, your extension will never be invoked. I would also suggest creating a new file in conf/dialplan/default instead of editing the default.xml file, unless of course you want to eliminate some of the extensions provided in default.xml in the sample configurations. Use the sofia status commands from fs_cli to see whether FreeSWITCH is registering to your gateways. for example, sofia status gateway If it doesn't work, the debug logs are in logs/freeswitch.log; read them carefully, as they will usually enable you to pinpoint the problem. To see where your extensions appear in the dial plan, have a look in logs/freeswitch.xml.fsxml, but don't edit that file! I hoe this helps. From diego.viola at gmail.com Sat Apr 11 23:30:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 12 Apr 2009 02:30:47 -0400 Subject: [Freeswitch-users] Question about play_and_get_digits in mod_dptools In-Reply-To: <86a32abc0904112047k2bc0df91ib6b6a2c703f6acd5@mail.gmail.com> References: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> <86a32abc0904111938o2e533481x6fdd0318940ca8a@mail.gmail.com> <86a32abc0904112047k2bc0df91ib6b6a2c703f6acd5@mail.gmail.com> Message-ID: <86a32abc0904112330q7454c12fl99d2c142ea3ec407@mail.gmail.com> I just added this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits Diego On Sat, Apr 11, 2009 at 11:47 PM, Diego Viola wrote: > Mike just answered. > > 23:45 < MikeJ> diegoviola: the syntax is documented > 23:45 < MikeJ> SWITCH_ADD_APP(app_interface, "play_and_get_digits", "Play > and get Digits", "Play and get Digits", > 23:45 < MikeJ> IIII play_and_get_digits_function, " > ", > SAF_NONE); > > Thanks guys. > > Diego > > > On Sat, Apr 11, 2009 at 10:38 PM, Diego Viola wrote: > >> If you give me some examples of how to use play_and_get_digits in >> mod_dptools I will document it here. >> >> >> http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_play_and_get_digits >> >> Diego >> >> >> On Sat, Apr 11, 2009 at 10:36 PM, Diego Viola wrote: >> >>> Hi all, >>> >>> I want to use play_and_get_digits from mod_dptools and have some >>> questions about it. >>> >>> I have used playAndGetDigits() in Lua but I see the syntax in mod_dptools >>> is a bit different and I got a bit confused. >>> >>> I see the syntax in the play_and_get_digits from the mod_dptools is >>> something like this: >>> >>> switch_play_and_get_digits(session, min_digits, max_digits, >>> max_tries, timeout, valid_terminators, >>> >>> prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, >>> sizeof(digit_buffer), digits_regex} >>> >>> Can you please explain to me what the session parameter is? And will this >>> allow me to use a phrase macro so I can call my IVR instead of calling a >>> regular file? >>> >>> This is how I use the playAndGetDigits in Lua: >>> >>> digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", >>> "", "\\d+"); >>> >>> I call a phrase macro instead of playing a file, can I do the same with >>> play_and_get_digits from mod_dptools? and please explain me what the session >>> parameter in play_and_get_digits is. >>> >>> Thanks, >>> >>> Diego >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090412/86a64435/attachment.html From diego.viola at gmail.com Sun Apr 12 00:50:32 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 12 Apr 2009 03:50:32 -0400 Subject: [Freeswitch-users] Question about play_and_get_digits in mod_dptools In-Reply-To: <86a32abc0904112330q7454c12fl99d2c142ea3ec407@mail.gmail.com> References: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> <86a32abc0904111938o2e533481x6fdd0318940ca8a@mail.gmail.com> <86a32abc0904112047k2bc0df91ib6b6a2c703f6acd5@mail.gmail.com> <86a32abc0904112330q7454c12fl99d2c142ea3ec407@mail.gmail.com> Message-ID: <86a32abc0904120050x4c6ddfe5hbf10f26be079b2d1@mail.gmail.com> I wish I would have seen this before =D 03:36 < Math> diegoviola: show application [appname] will show you any syntax btw Diego On Sun, Apr 12, 2009 at 2:30 AM, Diego Viola wrote: > I just added this: > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits > > Diego > > > On Sat, Apr 11, 2009 at 11:47 PM, Diego Viola wrote: > >> Mike just answered. >> >> 23:45 < MikeJ> diegoviola: the syntax is documented >> 23:45 < MikeJ> SWITCH_ADD_APP(app_interface, "play_and_get_digits", "Play >> and get Digits", "Play and get Digits", >> 23:45 < MikeJ> IIII play_and_get_digits_function, " >> ", >> SAF_NONE); >> >> Thanks guys. >> >> Diego >> >> >> On Sat, Apr 11, 2009 at 10:38 PM, Diego Viola wrote: >> >>> If you give me some examples of how to use play_and_get_digits in >>> mod_dptools I will document it here. >>> >>> >>> http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_play_and_get_digits >>> >>> Diego >>> >>> >>> On Sat, Apr 11, 2009 at 10:36 PM, Diego Viola wrote: >>> >>>> Hi all, >>>> >>>> I want to use play_and_get_digits from mod_dptools and have some >>>> questions about it. >>>> >>>> I have used playAndGetDigits() in Lua but I see the syntax in >>>> mod_dptools is a bit different and I got a bit confused. >>>> >>>> I see the syntax in the play_and_get_digits from the mod_dptools is >>>> something like this: >>>> >>>> switch_play_and_get_digits(session, min_digits, max_digits, >>>> max_tries, timeout, valid_terminators, >>>> >>>> prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, >>>> sizeof(digit_buffer), digits_regex} >>>> >>>> Can you please explain to me what the session parameter is? And will >>>> this allow me to use a phrase macro so I can call my IVR instead of calling >>>> a regular file? >>>> >>>> This is how I use the playAndGetDigits in Lua: >>>> >>>> digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", >>>> "", "\\d+"); >>>> >>>> I call a phrase macro instead of playing a file, can I do the same with >>>> play_and_get_digits from mod_dptools? and please explain me what the session >>>> parameter in play_and_get_digits is. >>>> >>>> Thanks, >>>> >>>> Diego >>>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090412/2bb937dc/attachment-0001.html From peter.olsson at visionutveckling.se Sun Apr 12 04:28:50 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 12 Apr 2009 13:28:50 +0200 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> Message-ID: Thanks, I tried latest trunk, but still no success.. :( Peter On 09-04-11 18.30, "Anthony Minessale" wrote: see if it works in latest trunk please. On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson wrote: When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem to detect DTMF. I'm not sure if this is a setting somewhere in the config files, or if it's a bug. The test scenario is simple - use the default FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't detect any DTMF tones. Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 e.schmidbauer at gmail.com Sun Apr 12 10:11:59 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Sun, 12 Apr 2009 13:11:59 -0400 Subject: [Freeswitch-users] ekiga and freeswitch Message-ID: <2cef777b0904121011q1e3df133xf2814d609effbbd7@mail.gmail.com> I am running FreeSWITCH Version 1.0.3 on a centos 5.2 x64 server and ekiga 3.2.0 on a client computer. I am able to register the ekiga client with the freeswitch server but when I dial an extension to join a conference I get the following error message: [ERR] switch_core_io.c:327 switch_core_session_read_frame() Codec RAW Signed Linear (16 bit) decoder error! I have the celt codec loaded on both the server and the client computer. This is the full output when dialing an extension: 2009-04-12 12:56:34 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing joseph schmidbauer->4800 in context default 2009-04-12 12:56:34 [INFO] mod_sofia.c:1301 sofia_receive_message() Asked to send early media by sofia/internal/joe at 192.168.1.125 2009-04-12 12:56:34 [INFO] mod_sofia.c:1342 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1239529036 1239529037 IN IP4 192.168.1.125 s=FreeSWITCH c=IN IP4 192.168.1.125 t=0 0 m=audio 26358 RTP/AVP 116 101 a=rtpmap:116 CELT/48000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv 2009-04-12 12:56:34 [NOTICE] mod_sofia.c:1345 sofia_receive_message() Pre-Answer sofia/internal/joe at 192.168.1.125! 2009-04-12 12:56:34 [NOTICE] mod_conference.c:1934 conference_loop_output() Channel [sofia/internal/joe at 192.168.1.125] has been answered warning: decode error [ERR] switch_core_io.c:327 switch_core_session_read_frame() Codec RAW Signed Linear (16 bit) decoder error! [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/internal/joe at 192.168.1.125 [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-12 13:05:14 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 4 (sofia/internal/joe at 192.168.1.125) Ended 2009-04-12 13:05:14 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/joe at 192.168.1.125 [CS_HANGUP] Not sure if this is a bug in the program or just in my setup. I've tried using the svn version of freeswitch (as of yesterday) and i got the exact same error. Any input would be appreciated. Thanks! From brian at freeswitch.org Sun Apr 12 10:34:29 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 12 Apr 2009 12:34:29 -0500 Subject: [Freeswitch-users] ekiga and freeswitch In-Reply-To: <2cef777b0904121011q1e3df133xf2814d609effbbd7@mail.gmail.com> References: <2cef777b0904121011q1e3df133xf2814d609effbbd7@mail.gmail.com> Message-ID: <159790A6-1FD5-4831-858B-47666E45C9B3@freeswitch.org> Collect a full sip trace and FULL console debug. Put it on our pastebin... Chances are Ekiga is doing something stupid... it usually does silly things. Also are you on SVN trunk? /b On Apr 12, 2009, at 12:11 PM, e schmidbauer wrote: > Not sure if this is a bug in the program or just in my setup. > I've tried using the svn version of freeswitch (as of yesterday) and i > got the exact same error. Any input would be appreciated. Thanks! From eric at rf.com Sun Apr 12 11:05:15 2009 From: eric at rf.com (Eric Chamberlain) Date: Sun, 12 Apr 2009 11:05:15 -0700 Subject: [Freeswitch-users] Can gateways be configured on a per user basis? Message-ID: <0FDF3663-4CC4-424B-99DE-143039A2213D@rf.com> Hello, I'm exploring the capabilities of FreeSwitch and have some questions: Can gateways be configured on a per user basis? A user needs to be limited to only their own gateways and other users can't user their gateway. Something like: User 1 has two gateways - VoIP provider A and VoIP provider B User 2 also has two gateways - VoIP provider A and VoIP provider C User 3 has ten gateways User 4 has one gateway etc. Is it possible to have some gateway settings global and some user specific, say proxy and port info global, but username and password are specific to each user? Is it also possible to specify on a per user basis whether the user's gateway will register for the provider? Using the example above, User 1 and User 2 would both register with VoIP provider A. And can that registration happen even if User 1 or User 2 don't have any SIP endpoints registered with FreeSwitch? -- Eric Chamberlain From nicolas at medularis.com Sun Apr 12 22:00:24 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Mon, 13 Apr 2009 01:00:24 -0400 Subject: [Freeswitch-users] Replace sqlite with couchDB? Message-ID: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> Hi, I am not very familiar with FS internals, but I recently found this "new" db engine called couchDB. Looks pretty interesting, and its main focus is scalability. Has anybody played with couchDB? does it make sense to replace sqlite with couchDB in FS? Here's a link to the project homepage: - http://couchdb.apache.org/ And here's a video of a presentation given by one of the lead programmers: - http://www.vimeo.com/1992869 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/6e89afdb/attachment.html From mattdfong at gmail.com Sun Apr 12 22:06:46 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Mon, 13 Apr 2009 12:06:46 +0700 Subject: [Freeswitch-users] Replace sqlite with couchDB? In-Reply-To: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> References: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> Message-ID: <4256bf830904122206v156a826dm43084ba30a315bd0@mail.gmail.com> Hi Nicolas, Just off the top of my head, but I think couchDB is rather large compared to sqlite, and I think it's also geared more towards storing dynamic datasets...rather ones that can be structured...like FS calling data can. But I might be wrong :) your buddy. --matt On Mon, Apr 13, 2009 at 12:00 PM, Nicolas Brenner wrote: > Hi, I am not very familiar with FS internals, but I recently found this > "new" db engine called couchDB. Looks pretty interesting, and its main focus > is scalability. > Has anybody played with couchDB? does it make sense to replace sqlite with > couchDB in FS? > > Here's a link to the project homepage: > - http://couchdb.apache.org/ > > And here's a video of a presentation given by one of the lead programmers: > - http://www.vimeo.com/1992869 > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/d382f8a5/attachment.html From nicolas at medularis.com Sun Apr 12 22:33:26 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Mon, 13 Apr 2009 01:33:26 -0400 Subject: [Freeswitch-users] Replace sqlite with couchDB? In-Reply-To: <4256bf830904122206v156a826dm43084ba30a315bd0@mail.gmail.com> References: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> <4256bf830904122206v156a826dm43084ba30a315bd0@mail.gmail.com> Message-ID: <1b46b4e80904122233j21201b2ela84c72fd1c597dcc@mail.gmail.com> Well, if it's too large compared to sqlite maybe it doesn't make sense. But I was thinking calling data is not always fixed. Depending on what you use FS for, you might want to get a CDR with many different data linked to each call, even different kinds of data linked to different calls, that would make each call very different and variable in its structure, which would fit a document db model. Thinking a bit more now, since couchdb is a document-based DB, it might be good for configuration-generating applications, like the ones consumed by xml_curl. These are external applications, yet they are still very closely related to FS, and might be able to benefit from using something like couchdb. On Mon, Apr 13, 2009 at 1:06 AM, Matthew Fong wrote: > Hi Nicolas, > Just off the top of my head, but I think couchDB is rather large compared > to sqlite, and I think it's also geared more towards > storing dynamic datasets...rather ones that can be structured...like FS > calling data can. > > But I might be wrong :) > your buddy. > > --matt > > On Mon, Apr 13, 2009 at 12:00 PM, Nicolas Brenner wrote: > >> Hi, I am not very familiar with FS internals, but I recently found this >> "new" db engine called couchDB. Looks pretty interesting, and its main focus >> is scalability. >> Has anybody played with couchDB? does it make sense to replace sqlite with >> couchDB in FS? >> >> Here's a link to the project homepage: >> - http://couchdb.apache.org/ >> >> And here's a video of a presentation given by one of the lead programmers: >> - http://www.vimeo.com/1992869 >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/8f87b080/attachment.html From jason at jasonjgw.net Sun Apr 12 22:52:36 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 13 Apr 2009 15:52:36 +1000 Subject: [Freeswitch-users] Replace sqlite with couchDB? In-Reply-To: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> References: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> Message-ID: <20090413055236.GA19344@jdc.jasonjgw.net> Nicolas Brenner wrote: > Hi, I am not very familiar with FS internals, but I recently found this > "new" db engine called couchDB. Looks pretty interesting, and its main focus > is scalability. > Has anybody played with couchDB? does it make sense to replace sqlite with > couchDB in FS? I think a lot of people would object to replacing a small database such as SQLite, which is easily integrated into the FreeSWITCh source code, with an Erlang application. Somehow, I don't see FreeSWITCH users accepting all of the dependencies that would bring, unless they're already using the Erlang module for other reasons. However, if it would be of benefit to Erlang users, I'm sure the FreeSWITCH developers would gladly accept a module. There are lots of databases out there, for example, http://monetdb.cwi.nl/ to mention just one that a Web search located for me. Which ones get supported depends on whether anyone is sufficiently interested to write modules for them. PostGRESQL and MySQL are already on the list, notably. From zhaoxxqq at 163.com Mon Apr 13 03:26:47 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Mon, 13 Apr 2009 18:26:47 +0800 Subject: [Freeswitch-users] Noise for dial out conference Message-ID: <200904131826464590579@163.com> Hello, I use below to realize call out conference: conference testconf bgdial {originate_timeout=30}sofia/default/1001 at 192.168.0.72 1234567890 FreeSWITCH_Conference" conference testconf bgdial {originate_timeout=30}sofia/default/1002 at 192.168.0.170 1234567890 FreeSWITCH_Conference" but when the second phone is connected. there are big noise in the phone, Can anyone help me to solve it? zhao xiaoqiang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/3c07e276/attachment.html From uv at yuvalhertzog.com Mon Apr 13 04:32:12 2009 From: uv at yuvalhertzog.com (UV) Date: Mon, 13 Apr 2009 21:32:12 +1000 Subject: [Freeswitch-users] Skypiax as a windows service Message-ID: <0D1E9E22CCAC4F98ADA9863FDFF7FB85@UVix> Great work on Skypiax, Giovanni. We've tested it in our lab for sometime and it works very well. Unfortunately, when we tried deploying it on a production environment (running Win2K3 server farm), we ran into a barrier: 1. FS is running as terminal server console application (to be easily maintained remotely by RDP) 2. This is because Win2K does not allow RDP to access system console (session /userid 0) 3. Skype does not work on terminal server due to a well known disappearing audio drivers problem, therefore it has to run either as a console or a service (both on session 0). 4. FS can run well as a windows service 5. Skypiax seem to load as service, but it can't find the skype client and exit with the following error: 2009-04-13 20:54:14 [ERR] mod_skypiax.c:990 load_config() rev 13006M[00000000|37 ][ERRORA 990 ][skype_user ][-1, 0, 0] Failed to connect to a SKYPE API for interface_id=1, no SKYPE client running, please (re)start Skype client. Skypiax exiting This situation prevents me to run skypiax in production. I understand from the wiki page that windows service is not done yet - so I presume this is a predicted outcome. Any idea when and if this is planned to be implemented? Keep up the good work! Cheers, UV -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/a20a7842/attachment.html From yudha2008 at gmail.com Mon Apr 13 06:09:38 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 13 Apr 2009 18:39:38 +0530 Subject: [Freeswitch-users] Mod_java loading error Message-ID: Hi, I have loaded the java module in freeswitch. But when i run freeswitch in console i get this error. 2009-04-13 17:38:33 [NOTICE] modjava.c:244 mod_java_load() Java Framework Loading... 2009-04-13 17:38:33 [ERR] modjava.c:133 load_config() Error loading /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so 2009-04-13 17:38:33 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** Can any one specify what is the error. I am using Freeswitch 1.0.2 in CentOS 5.2. Thanks in advance. -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/608abf4b/attachment.html From brian at freeswitch.org Mon Apr 13 06:15:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 13 Apr 2009 08:15:39 -0500 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: Message-ID: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> You should update to SVN trunk and try again. /b On Apr 13, 2009, at 8:09 AM, Baskar wrote: > Hi, > > I have loaded the java module in freeswitch. But when i run > freeswitch in console i get this error. > > 2009-04-13 17:38:33 [NOTICE] modjava.c:244 mod_java_load() Java > Framework Loading... > 2009-04-13 17:38:33 [ERR] modjava.c:133 load_config() Error loading / > usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so > 2009-04-13 17:38:33 [CRIT] switch_loadable_module.c:839 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_java.so > **Module load routine returned an error** > > Can any one specify what is the error. I am using Freeswitch 1.0.2 > in CentOS 5.2. > > Thanks in advance. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/3bacc781/attachment.html From grevenx at me.com Mon Apr 13 08:13:39 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Mon, 13 Apr 2009 17:13:39 +0200 Subject: [Freeswitch-users] Replace sqlite with couchDB? In-Reply-To: <20090413055236.GA19344@jdc.jasonjgw.net> References: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> <20090413055236.GA19344@jdc.jasonjgw.net> Message-ID: <92AD0C5B-366C-4947-B6ED-9F10C6EBADB8@me.com> The only part I see fit for integration with CouchDB is for storing CDR documents. This kind of database is imho best-used for storing large sets of data, in a document structure. I don't think the FS config fits this description, since the amount of config documents are typically not "large". You can also look at related distributed systems like Hadoop/Hbase, which could be good to store CDRs in. It's been a couple of months since I researched these systems, but I think it's possible to enable an HTTP REST interface for both, so you could use the built-in feature for posting CDRs to a HTTP server. Best regards, Even Andr? On 13. april. 2009, at 07.52, Jason White wrote: > Nicolas Brenner wrote: >> Hi, I am not very familiar with FS internals, but I recently found >> this >> "new" db engine called couchDB. Looks pretty interesting, and its >> main focus >> is scalability. >> Has anybody played with couchDB? does it make sense to replace >> sqlite with >> couchDB in FS? > > I think a lot of people would object to replacing a small database > such as > SQLite, which is easily integrated into the FreeSWITCh source code, > with an > Erlang application. Somehow, I don't see FreeSWITCH users accepting > all of the > dependencies that would bring, unless they're already using the > Erlang module > for other reasons. However, if it would be of benefit to Erlang > users, I'm > sure the FreeSWITCH developers would gladly accept a module. > > There are lots of databases out there, for example, http://monetdb.cwi.nl/ > to > mention just one that a Web search located for me. > > Which ones get supported depends on whether anyone is sufficiently > interested > to write modules for them. > > PostGRESQL and MySQL are already on the list, notably. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Mon Apr 13 08:25:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Apr 2009 10:25:22 -0500 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: References: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> Message-ID: <191c3a030904130825q32f59ddagf19700aea64e5dfe@mail.gmail.com> which rev was "latest" for you? I have confirmation that it is indeed working. On Sun, Apr 12, 2009 at 6:28 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Thanks, > > I tried latest trunk, but still no success.. :( > > Peter > > > On 09-04-11 18.30, "Anthony Minessale" > wrote: > > see if it works in latest trunk please. > > > On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem > to detect DTMF. I'm not sure if this is a setting somewhere in the config > files, or if it's a bug. The test scenario is simple - use the default > FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't > detect any DTMF tones. > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/11bd27c9/attachment-0001.html From peter.olsson at visionutveckling.se Mon Apr 13 08:29:59 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 13 Apr 2009 17:29:59 +0200 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... Message-ID: I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson From anthony.minessale at gmail.com Mon Apr 13 08:33:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Apr 2009 10:33:08 -0500 Subject: [Freeswitch-users] Can gateways be configured on a per user basis? In-Reply-To: <0FDF3663-4CC4-424B-99DE-143039A2213D@rf.com> References: <0FDF3663-4CC4-424B-99DE-143039A2213D@rf.com> Message-ID: <191c3a030904130833hf3c73a5meca407f98f77b6d0@mail.gmail.com> You can store gateway xml in both a user and in the sofia.conf, we don't enforce which ones who can use because that would be a function of the dialplan but you can certainly store them in your configs that way and set the names of a user's specifc gateways in a variable that would be present on each inbound call. On Sun, Apr 12, 2009 at 1:05 PM, Eric Chamberlain wrote: > Hello, > > I'm exploring the capabilities of FreeSwitch and have some questions: > > Can gateways be configured on a per user basis? > > A user needs to be limited to only their own gateways and other users > can't user their gateway. > > Something like: > > User 1 has two gateways - VoIP provider A and VoIP provider B > User 2 also has two gateways - VoIP provider A and VoIP provider C > User 3 has ten gateways > User 4 has one gateway > etc. > > Is it possible to have some gateway settings global and some user > specific, say proxy and port info global, but username and password > are specific to each user? > > Is it also possible to specify on a per user basis whether the user's > gateway will register for the provider? Using the example above, User > 1 and User 2 would both register with VoIP provider A. > And can that registration happen even if User 1 or User 2 don't have > any SIP endpoints registered with FreeSwitch? > > -- > Eric Chamberlain > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/f6a574e9/attachment.html From peter.olsson at visionutveckling.se Mon Apr 13 08:42:56 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 13 Apr 2009 17:42:56 +0200 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: <191c3a030904130825q32f59ddagf19700aea64e5dfe@mail.gmail.com> Message-ID: Anthony, Please read my comments on Jira cases MODOPAL-3 and MODOPAL-5. In MODOPAL-3 I've also attached a patch to get DTMF working better (handle both Tone and String input), and also increasing the stability of mod_opal. But the problem also is connected to MODOPAL-5, which I created just now. It seems to be a codec issue when using A-Law, which I didn't found out until now - that's why the in-band DTMF detection didn't work. But I have DTMF working right now, but it's a bit improved with my patch :) Regards, Peter On 09-04-13 17.25, "Anthony Minessale" wrote: which rev was "latest" for you? I have confirmation that it is indeed working. On Sun, Apr 12, 2009 at 6:28 AM, Peter Olsson wrote: Thanks, I tried latest trunk, but still no success.. :( Peter On 09-04-11 18.30, "Anthony Minessale" wrote: see if it works in latest trunk please. On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson wrote: When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem to detect DTMF. I'm not sure if this is a setting somewhere in the config files, or if it's a bug. The test scenario is simple - use the default FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't detect any DTMF tones. Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 Mon Apr 13 08:54:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Apr 2009 10:54:56 -0500 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: References: Message-ID: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> 1) When you say latest, which rev does that mean? we change revs pretty often. 2) Do you have a minimal script that reproduces your issue. 3) is there a reason you cannot just session.execute("bridge", dest); instead of doing it manually (which is a process not for the faint at heart)? On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > I have two problems that I haven't been able to solve. I've done the same > tests in both javascript, and in .NET. > > The two scripts are pretty simple, they just answer an incomming call, > creates a new session, wait for an answer on the second call leg, and then > bridge the two channels together. > > In both cases everything works just fine, but the audio is distorted. The > destination I'm calling is "loopback/5000" - the sample IVR application > included in FreeSWITCH. I first thought it was a codec issue, but even after > trying to switch to different codecs the problem was the same. It more > sounds like it's a timestamping issue - the voice is not distorted enough to > be a bad codec, but it reads way to fast (mayby twice the "normal" speed). > When doing a direct transfer() to the other destination this works just > fine, but I need to be able to have some extra logic to tell if the > destination is available or not. > > The second problem occurs only in .NET. After doing this sample there is as > loopback channel still hanging around. It seems like the call creates a > loopback-a and loopback-b, the loopback-b dissapears as it should (when the > call has been disconnected), but the other one stays there. When doing the > same in javascript this doesn't seem to occur. > > I'm using the latest SVN trunk, and my OS is Windows XP. > > I found bug FSCORE-349 in Jira, which seems to point in to the direction > that there might be a bug with the loopback channels in some cases, but I > could not find anything about the audio which plays too fast. > > Has anyone else experienced this? > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/21fe407e/attachment.html From peter.olsson at visionutveckling.se Mon Apr 13 09:21:22 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 13 Apr 2009 18:21:22 +0200 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> Message-ID: 1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available.. 2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg. 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. :) What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected? --- if (session.ready()) { session.answer(); new_session = new Session("loopback/5000", session); new_session.waitForAnswer(); bridge(session, new_session); // Not sure if this is needed - I've tried with it both enabled and disabled session.hangup(); new_session.hangup(); } Peter On 09-04-13 17.54, "Anthony Minessale" wrote: 1) When you say latest, which rev does that mean? we change revs pretty often. 2) Do you have a minimal script that reproduces your issue. 3) is there a reason you cannot just session.execute("bridge", dest); instead of doing it manually (which is a process not for the faint at heart)? On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson wrote: I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 berni at birkenwald.de Mon Apr 13 09:35:06 2009 From: berni at birkenwald.de (Bernhard Schmidt) Date: Mon, 13 Apr 2009 16:35:06 +0000 (UTC) Subject: [Freeswitch-users] Can't call registered user in internal-ipv6 profile Message-ID: Hi, probably a pretty easy problem, but I can't figure it out nevertheless. I'm still experimenting with FreeSwitch (SVN trunk, about two weeks old), mainly due to the IPv6 SIP support. I'm pretty much running the default configuration included in the sourcetree. Now I've hit the following problem: I have two phones registered, one Snom with extension 1000 on IPv4 (profile internal), one SIP Communicator with extension 1002 on IPv6 (profile internal-ipv6). I can call the Snom just fine (from the SIP Communicator as well as from outside or the CLI), but not the SIP Communicator. EXECUTE sofia/internal/1000 at obelix.oms16.birkenwald.de bridge(user/1002 at 172.16.1.69) 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:1077 switch_ivr_originate() variable string 0 = [presence_id=1002 at 172.16.1.69] 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 switch_ivr_originate() Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 switch_ivr_originate() Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 switch_ivr_originate() Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [INFO] mod_dptools.c:2051 audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED It works when I explicitly specify the internal-ipv6 profile for the outgoing call like this: but that isn't really what I want. What are the quirks I need to add to have "user/@" search both profiles? I already set force-register-domain on the profile, but I don't think that is what I'm looking for. Bernhard From fialkam at gmail.com Mon Apr 13 09:56:11 2009 From: fialkam at gmail.com (Martin Fiala) Date: Mon, 13 Apr 2009 18:56:11 +0200 Subject: [Freeswitch-users] SIP switching made simple? Message-ID: Hello. I'm trying to use freeswitch, was able to compile it without problems, which is very nice. Then studying the configurations etc., I managed to set up SIP accounts those register properly. But now, if I want to call one registered account from the other one, I get error 404 - not found. I tried to set up a minimalistic dialplan using xml syntax as well as asterisk syntax but neither worked for me. I changed just a few thing, I'll list them later. I'm trying to make calls using ip addresses and ports instead of domain names.. This is the error freeswitch outputs: 2009-04-13 18:35:48 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/02 at 192.168.2.100 [19bad83a-ec9a-4b59-8457-cd76f1eaef65] 2009-04-13 18:35:48 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 02->01 in context default 2009-04-13 18:35:48 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/02 at 192.168.2.100 to enum[01 at default] 2009-04-13 18:35:50 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting 2009-04-13 18:35:50 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup sofia/internal/02 at 192.168.2.100 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 1 (sofia/internal/02 at 192.168.2.100) Ended 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/02 at 192.168.2.100 [CS_HANGUP] My users are added in file users.xml in directory/ : I've added the file dialplan/default/000_default.xml with contents: That's from sample configs, I wonder, if the IP address can be used like that. I understand it that way, the ip address specified is of registrar server. I've added the port as I'm testing it on local loop and thus am running different sip services on the same ip (freeswitch and calling softfones). Is that ok? extensions.conf I've tried to use: [default] ; Things you're used to.... ;exten => music,n,Dial(SIP/1234 at conference.freeswitch.org|120) ;exten => _1XXXXX,n,set(cool=${EXTEN}) ;exten => _1XXXXX,n,set(myvar=true) ;exten => _1XXXXX,n,Goto(default|music) ;exten => 2137991400/1000,n,Goto(default|music) ; Some new magic you can do.... ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,enum($1) ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,bridge(${enum_auto_route}) ; instead of exten, put anything about the call you would rather match on. ; either the names of a field in caller_profile or a string of variables to expand. ;caller_id_number => 2137991400,n,Goto(default|music) ;${sip_from_user} => bill,n,Goto(default|music) [pbx] exten => 01,1,Dial(SIP/01,20) exten => 02,1,Dial(SIP/02,20) When using extensions.conf I've changed this line in sip_profiles/internal.xml from: to I didn't make any other changes in that file. I didn't change anything else. I'm trying to use two sip phones - one using port 6001 (user "01") and the other one 5000 (user "02"). After registration succeeds, calling this sip uri : sip:01 at 192.168.2.100:5060, where 192.168.2.100:5060 is IP:PORT of freeswitch (the IP is same for softphones.. the same machine). Thanks for any help. Fiala From msc at freeswitch.org Mon Apr 13 10:13:27 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Apr 2009 10:13:27 -0700 Subject: [Freeswitch-users] Noise for dial out conference In-Reply-To: <200904131826464590579@163.com> References: <200904131826464590579@163.com> Message-ID: <87f2f3b90904131013y1de87de8v5ab9679d27f68df7@mail.gmail.com> Are both of these phones on the same LAN as FreeSWITCH? What kind of phones are they? Also, can you reverse the dialing order and reproduce the symptoms? Just curious if it's always the same phone causing the noise or if it is always the second phone connected, regardless of which phone. -MC On Mon, Apr 13, 2009 at 3:26 AM, zhaoxxqq wrote: > Hello, > > I use below to realize call out conference: > conference testconf bgdial {originate_timeout=30}sofia/default/ > 1001 at 192.168.0.72 1234567890 FreeSWITCH_Conference" > conference testconf bgdial {originate_timeout=30}sofia/default/ > 1002 at 192.168.0.170 1234567890 FreeSWITCH_Conference" > but when the second phone is connected. there are big noise in the phone, > Can anyone help me to solve it? > > zhao xiaoqiang > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/91ff7ff2/attachment.html From brian at freeswitch.org Mon Apr 13 10:17:29 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 13 Apr 2009 12:17:29 -0500 Subject: [Freeswitch-users] Can't call registered user in internal-ipv6 profile In-Reply-To: References: Message-ID: <04D731AC-5A70-4515-8ABA-B432D18871FA@freeswitch.org> user/ uses the dial-string in the domain to find the user on the internal profile by default.. so to call someone registered via ipv6 you'll need to put a dial-string param in the user to find them on the internal-ipv6 profile. /b On Apr 13, 2009, at 11:35 AM, Bernhard Schmidt wrote: > Hi, > > probably a pretty easy problem, but I can't figure it out > nevertheless. > > I'm still experimenting with FreeSwitch (SVN trunk, about two weeks > old), mainly due to the IPv6 SIP support. I'm pretty much running the > default configuration included in the sourcetree. Now I've hit the > following problem: > > I have two phones registered, one Snom with extension 1000 on IPv4 > (profile internal), one SIP Communicator with extension 1002 on IPv6 > (profile internal-ipv6). I can call the Snom just fine (from the SIP > Communicator as well as from outside or the CLI), but not the SIP > Communicator. > > EXECUTE sofia/internal/1000 at obelix.oms16.birkenwald.de bridge(user/1002 at 172.16.1.69 > ) > 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:1077 > switch_ivr_originate() variable string 0 = [presence_id=1002 at 172.16.1.69 > ] > 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 > switch_ivr_originate() Cannot create outgoing channel of type > [error] cause: [USER_NOT_REGISTERED] > 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 > switch_ivr_originate() Originate Resulted in Error Cause: 606 > [USER_NOT_REGISTERED] > 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 > switch_ivr_originate() Cannot create outgoing channel of type [user] > cause: [USER_NOT_REGISTERED] > 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 > switch_ivr_originate() Originate Resulted in Error Cause: 606 > [USER_NOT_REGISTERED] > 2009-04-13 18:23:45 [INFO] mod_dptools.c:2051 > audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED > > It works when I explicitly specify the internal-ipv6 profile for the > outgoing call like this: > > > > but that isn't really what I want. > > What are the quirks I need to add to have "user/@" > search both profiles? I already set force-register-domain on the > profile, but I don't think that is what I'm looking for. > > Bernhard > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/0a14f6f6/attachment.html From msc at freeswitch.org Mon Apr 13 10:25:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Apr 2009 10:25:16 -0700 Subject: [Freeswitch-users] SIP switching made simple? In-Reply-To: References: Message-ID: <87f2f3b90904131025k635be5faqa46cc9228381aedd@mail.gmail.com> I highly recommend that you set aside this endeavor for the time being and use the default configuration. Once you get familiar with the default config then you'll realize how to make changes to registered users and to the dialplan. Don't let the size of the default configuration scare you off. It is very well designed, and much of it is compartmentalized, which means you can changes in a single file without affecting the rest of the configuration. Now for the usual questions: What platform are you on? Linux? Did you use SVN? (We highly recommend using SVN) Have you seen the wiki pages on installing FS? If you're running on Linux then I recommend a clean install using the method documented here: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install Let us know how it goes. -MC On Mon, Apr 13, 2009 at 9:56 AM, Martin Fiala wrote: > Hello. > > I'm trying to use freeswitch, was able to compile it without problems, > which is very nice. Then studying the configurations etc., I managed > to set up SIP accounts those register properly. But now, if I want to > call one registered account from the other one, I get error 404 - not > found. I tried to set up a minimalistic dialplan using xml syntax as > well as asterisk syntax but neither worked for me. I changed just a > few thing, I'll list them later. I'm trying to make calls using ip > addresses and ports instead of domain names.. > > This is the error freeswitch outputs: > 2009-04-13 18:35:48 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel sofia/internal/02 at 192.168.2.100 > [19bad83a-ec9a-4b59-8457-cd76f1eaef65] > 2009-04-13 18:35:48 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 02->01 in context default > 2009-04-13 18:35:48 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() Transfer sofia/internal/02 at 192.168.2.100 > to enum[01 at default] > 2009-04-13 18:35:50 [INFO] switch_core_state_machine.c:122 > switch_core_standard_on_routing() No Route, Aborting > 2009-04-13 18:35:50 [NOTICE] switch_core_state_machine.c:123 > switch_core_standard_on_routing() Hangup > sofia/internal/02 at 192.168.2.100 [CS_ROUTING] [NO_ROUTE_DESTINATION] > 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 1 > (sofia/internal/02 at 192.168.2.100) Ended > 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel > sofia/internal/02 at 192.168.2.100 [CS_HANGUP] > > My users are added in file users.xml in directory/ : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I've added the file dialplan/default/000_default.xml with contents: > > > > /> > > > That's from sample configs, I wonder, if the IP address can be used > like that. I understand it that way, the ip address specified is of > registrar server. I've added the port as I'm testing it on local loop > and thus am running different sip services on the same ip (freeswitch > and calling softfones). Is that ok? > > > > extensions.conf I've tried to use: > [default] > > ; Things you're used to.... > ;exten => music,n,Dial(SIP/1234 at conference.freeswitch.org|120) > > ;exten => _1XXXXX,n,set(cool=${EXTEN}) > ;exten => _1XXXXX,n,set(myvar=true) > ;exten => _1XXXXX,n,Goto(default|music) > ;exten => 2137991400/1000,n,Goto(default|music) > > > ; Some new magic you can do.... > ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,enum($1) > ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,bridge(${enum_auto_route}) > > ; instead of exten, put anything about the call you would rather match on. > ; either the names of a field in caller_profile or a string of > variables to expand. > ;caller_id_number => 2137991400,n,Goto(default|music) > ;${sip_from_user} => bill,n,Goto(default|music) > > [pbx] > exten => 01,1,Dial(SIP/01,20) > exten => 02,1,Dial(SIP/02,20) > > > > > When using extensions.conf I've changed this line in > sip_profiles/internal.xml from: > > to > > I didn't make any other changes in that file. > > > I didn't change anything else. > > I'm trying to use two sip phones - one using port 6001 (user "01") > and the other one 5000 (user "02"). After registration succeeds, > calling this sip uri : sip:01 at 192.168.2.100:5060, where > 192.168.2.100:5060 is IP:PORT of freeswitch (the IP is same for > softphones.. the same machine). > > Thanks for any help. > Fiala > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/b659f281/attachment-0001.html From anthony.minessale at gmail.com Mon Apr 13 11:37:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Apr 2009 13:37:43 -0500 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> Message-ID: <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> see how it works in latest trunk 13011 nontheless you can just say session.execute("bridge", "loopback/5000"); and get the same result without touching that other channel. when the call fails, you will have an originate_disposition variable in session you can check. On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > 1. The latest trunk I've tried with is 13008. Since I'm not doing > anything for production yet (just testing/evaluating), so I tend to update > as soon as there is new version available.. > 2. Yep, you will find it below. In javascript - my sample for .NET does > basically the same thing, with the same result, except that it also won't > drop the loopback-a call leg. > 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess > I'm not 100% sure what I'm doing.. :) What I want to be able to do is to > dial into a script, let the script dial another extension, and bridge them > together when the other party answers the call. I also need to take care of > call setup problems - if the other part doesn't respond, is unavailable or > busy in the phone - so I though this was the only way? If I use the > session.execute("bridge"..), will I be able to control the call if it > couldn't be connected? > > --- > > if (session.ready()) { > > session.answer(); > > new_session = new Session("loopback/5000", session); > new_session.waitForAnswer(); > > bridge(session, new_session); > > // Not sure if this is needed - I've tried with it both enabled and > disabled > session.hangup(); > new_session.hangup(); > } > > Peter > > > On 09-04-13 17.54, "Anthony Minessale" > wrote: > > 1) When you say latest, which rev does that mean? we change revs pretty > often. > 2) Do you have a minimal script that reproduces your issue. > 3) is there a reason you cannot just session.execute("bridge", dest); > instead of doing it manually (which is a process not for the faint at > heart)? > > > > On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > I have two problems that I haven't been able to solve. I've done the same > tests in both javascript, and in .NET. > > The two scripts are pretty simple, they just answer an incomming call, > creates a new session, wait for an answer on the second call leg, and then > bridge the two channels together. > > In both cases everything works just fine, but the audio is distorted. The > destination I'm calling is "loopback/5000" - the sample IVR application > included in FreeSWITCH. I first thought it was a codec issue, but even after > trying to switch to different codecs the problem was the same. It more > sounds like it's a timestamping issue - the voice is not distorted enough to > be a bad codec, but it reads way to fast (mayby twice the "normal" speed). > When doing a direct transfer() to the other destination this works just > fine, but I need to be able to have some extra logic to tell if the > destination is available or not. > > The second problem occurs only in .NET. After doing this sample there is as > loopback channel still hanging around. It seems like the call creates a > loopback-a and loopback-b, the loopback-b dissapears as it should (when the > call has been disconnected), but the other one stays there. When doing the > same in javascript this doesn't seem to occur. > > I'm using the latest SVN trunk, and my OS is Windows XP. > > I found bug FSCORE-349 in Jira, which seems to point in to the direction > that there might be a bug with the loopback channels in some cases, but I > could not find anything about the audio which plays too fast. > > Has anyone else experienced this? > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/32e37184/attachment.html From berni at birkenwald.de Mon Apr 13 12:04:26 2009 From: berni at birkenwald.de (Bernhard Schmidt) Date: Mon, 13 Apr 2009 19:04:26 +0000 (UTC) Subject: [Freeswitch-users] Can't call registered user in internal-ipv6 profile References: <04D731AC-5A70-4515-8ABA-B432D18871FA@freeswitch.org> Message-ID: Brian West wrote: > user/ uses the dial-string in the domain to find the user on the > internal profile by default.. so to call someone registered via ipv6 > you'll need to put a dial-string param in the user to find them on the > internal-ipv6 profile. Oh okay, that works, thanks a lot. The default is from directory/default.xml, right? How does that point to the internal profile? Or is this choice hardcoded somewhere? Is there any pitfall replacing the default with it works, but I think there are a lot of things in Freeswitch I haven't fully understood yet. Regards, Bernhard >> probably a pretty easy problem, but I can't figure it out >> nevertheless. >> >> I'm still experimenting with FreeSwitch (SVN trunk, about two weeks >> old), mainly due to the IPv6 SIP support. I'm pretty much running the >> default configuration included in the sourcetree. Now I've hit the >> following problem: >> >> I have two phones registered, one Snom with extension 1000 on IPv4 >> (profile internal), one SIP Communicator with extension 1002 on IPv6 >> (profile internal-ipv6). I can call the Snom just fine (from the SIP >> Communicator as well as from outside or the CLI), but not the SIP >> Communicator. >> >> EXECUTE sofia/internal/1000 at obelix.oms16.birkenwald.de bridge(user/1002-BnVRy6f7ncgPVn3RC9QTCQ at public.gmane.org >> ) >> 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:1077 >> switch_ivr_originate() variable string 0 = [presence_id@?Hak?T??8????? >> ] >> 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 >> switch_ivr_originate() Cannot create outgoing channel of type >> [error] cause: [USER_NOT_REGISTERED] >> 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 >> switch_ivr_originate() Originate Resulted in Error Cause: 606 >> [USER_NOT_REGISTERED] >> 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 >> switch_ivr_originate() Cannot create outgoing channel of type [user] >> cause: [USER_NOT_REGISTERED] >> 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 >> switch_ivr_originate() Originate Resulted in Error Cause: 606 >> [USER_NOT_REGISTERED] >> 2009-04-13 18:23:45 [INFO] mod_dptools.c:2051 >> audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED >> >> It works when I explicitly specify the internal-ipv6 profile for the >> outgoing call like this: >> >> >> >> but that isn't really what I want. >> >> What are the quirks I need to add to have "user/@" >> search both profiles? I already set force-register-domain on the >> profile, but I don't think that is what I'm looking for. From diego.viola at gmail.com Mon Apr 13 22:36:09 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 01:36:09 -0400 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools Message-ID: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> Hi everyone, I have a question... I have this on my dialplan: What I want to do is play and read some digits and as soon as I get those digits, transfer to that extension... but this never happens, even if I terminate with a #. I do the same thing with Lua and it works with Lua, but I need it to work with play_and_get_digits from mod_dptools, because I plan to use this with event socket outbound, with an application which I'm currently working on. Any ideas? Thanks, Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/b9cfe5f8/attachment.html From diego.viola at gmail.com Mon Apr 13 22:42:45 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 01:42:45 -0400 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools In-Reply-To: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> References: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> Message-ID: <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> It works if I use "read" and do this: But I need play_and_get_digits to work like that too, please. Diego On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote: > Hi everyone, > > I have a question... I have this on my dialplan: > > > > > > > > > What I want to do is play and read some digits and as soon as I get those > digits, transfer to that extension... but this never happens, even if I > terminate with a #. > > I do the same thing with Lua and it works with Lua, but I need it to work > with play_and_get_digits from mod_dptools, because I plan to use this with > event socket outbound, with an application which I'm currently working on. > > Any ideas? > > Thanks, > > Diego > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/ac771dc3/attachment.html From diego.viola at gmail.com Mon Apr 13 22:53:39 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 01:53:39 -0400 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools In-Reply-To: <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> References: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> Message-ID: <86a32abc0904132253r57a4a9d3mf0f372473544276f@mail.gmail.com> I remember playAndGetDigits had a bug like this too. Anthony, please help me. On Tue, Apr 14, 2009 at 1:42 AM, Diego Viola wrote: > It works if I use "read" and do this: > > > > > > > > > > > But I need play_and_get_digits to work like that too, please. > > Diego > > > On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote: > >> Hi everyone, >> >> I have a question... I have this on my dialplan: >> >> >> >> >> >> >> >> >> What I want to do is play and read some digits and as soon as I get those >> digits, transfer to that extension... but this never happens, even if I >> terminate with a #. >> >> I do the same thing with Lua and it works with Lua, but I need it to work >> with play_and_get_digits from mod_dptools, because I plan to use this with >> event socket outbound, with an application which I'm currently working on. >> >> Any ideas? >> >> Thanks, >> >> Diego >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/332280d3/attachment-0001.html From moizchinoy at gmail.com Mon Apr 13 23:02:29 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Tue, 14 Apr 2009 10:02:29 +0400 Subject: [Freeswitch-users] Google Talk Integration... Message-ID: <29b888f80904132302k22e45594w997f70bd48be28e2@mail.gmail.com> Hi, I have tried google talk integration with FS and is working fine. Great Work! Is it possible to have multiple concurrent incoming calls on the same gmail account? -- Regards, Moiz Chinoy. From diego.viola at gmail.com Mon Apr 13 23:02:46 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 02:02:46 -0400 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools In-Reply-To: <86a32abc0904132253r57a4a9d3mf0f372473544276f@mail.gmail.com> References: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> <86a32abc0904132253r57a4a9d3mf0f372473544276f@mail.gmail.com> Message-ID: <86a32abc0904132302y41981bd9w11f6a4c9fac5b5c3@mail.gmail.com> Anthony, I just tried to print the variable with the log app, with read it prints, with play_and_get_digits doesn't. I'm using latest SVN rev: FreeSWITCH Version 1.0.trunk (13012M) Thanks, Diego On Tue, Apr 14, 2009 at 1:53 AM, Diego Viola wrote: > I remember playAndGetDigits had a bug like this too. > > Anthony, please help me. > > > On Tue, Apr 14, 2009 at 1:42 AM, Diego Viola wrote: > >> It works if I use "read" and do this: >> >> >> >> >> >> >> >> >> >> >> But I need play_and_get_digits to work like that too, please. >> >> Diego >> >> >> On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote: >> >>> Hi everyone, >>> >>> I have a question... I have this on my dialplan: >>> >>> >>> >>> >>> >>> >>> >>> >>> What I want to do is play and read some digits and as soon as I get those >>> digits, transfer to that extension... but this never happens, even if I >>> terminate with a #. >>> >>> I do the same thing with Lua and it works with Lua, but I need it to work >>> with play_and_get_digits from mod_dptools, because I plan to use this with >>> event socket outbound, with an application which I'm currently working on. >>> >>> Any ideas? >>> >>> Thanks, >>> >>> Diego >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/8d66df9e/attachment.html From yudha2008 at gmail.com Tue Apr 14 00:24:48 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 14 Apr 2009 12:54:48 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> Message-ID: *Hi Brian West,* * I have installed the latest SVN Freeswitch trunk but still i get the same error. How can i over come this problem. 2009-04-14 12:44:26 [NOTICE] modjava.c:244 mod_java_load() Java Framework Loading... 2009-04-14 12:44:26 [ERR] modjava.c:133 load_config() Error loading /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so 2009-04-14 12:44:26 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** *2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_lua] 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:248 switch_loadable_module_process() Adding Application 'lua' 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'luarun' 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'lua' 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_say_en] 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:395 switch_loadable_module_process() Adding Say interface 'en' 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:120 switch_loadable_module_runtime() Starting runtime thread for CORE_SOFTTIMER_MODULE 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:120 switch_loadable_module_runtime() Starting runtime thread for mod_event_socket 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 switch_load_network_lists() Created ip list dl-candidates default (allow) 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 10.0.0.0/8 (deny) to list dl-candidates 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 172.16.0.0/12 (deny) to list dl-candidates 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 192.168.0.0/16 (deny) to list dl-candidates 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 switch_load_network_lists() Created ip list rfc1918 default (deny) 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 10.0.0.0/8 (allow) to list rfc1918 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 172.16.0.0/12 (allow) to list rfc1918 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 192.168.0.0/16 (allow) to list rfc1918 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 switch_load_network_lists() Created ip list lan default (allow) 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 192.168.42.0/24 (deny) to list lan 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 192.168.42.42/32 (allow) to list lan 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 switch_load_network_lists() Created ip list strict default (deny) 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 208.102.123.124/32 (allow) to list strict 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 switch_load_network_lists() Created ip list domains default (deny) 2009-04-14 12:44:26 [NOTICE] switch_core.c:965 switch_load_network_lists() Adding 192.0.2.0/24 (allow) [brian at 192.168.1.140] to list domains 2009-04-14 12:44:26 [CONSOLE] switch_core.c:1322 switch_core_init_and_modload() *FreeSWITCH Version 1.0.trunk (13013M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] Specify what is the error why i cant able to load Mod_java. -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/093cbce8/attachment.html From solko at gcdf.pl Tue Apr 14 00:59:40 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 14 Apr 2009 09:59:40 +0200 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> Message-ID: <49E4426C.6010400@gcdf.pl> Baskar pisze: > *Hi Brian West,* > > > * I have installed the latest SVN Freeswitch trunk but still i get the > same error. How can i over come this problem. > > 2009-04-14 12:44:26 [NOTICE] modjava.c:244 mod_java_load() Java > Framework Loading... > 2009-04-14 12:44:26 [ERR] modjava.c:133 load_config() Error loading > /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so First of all do you have java installed in that path? If not edit configuration path. I do not use java mod for some time but I had no problem to load them, only reloading was a problem. > 2009-04-14 12:44:26 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_java.so > **Module load routine returned an error** > *2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:889 > switch_loadable_module_load_file() Successfully Loaded [mod_lua] > 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:248 > switch_loadable_module_process() Adding Application 'lua' > 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:270 > switch_loadable_module_process() Adding API Function 'luarun' > 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:270 > switch_loadable_module_process() Adding API Function 'lua' > 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:889 > switch_loadable_module_load_file() Successfully Loaded [mod_say_en] > 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:395 > switch_loadable_module_process() Adding Say interface 'en' > 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:120 > switch_loadable_module_runtime() Starting runtime thread for > CORE_SOFTTIMER_MODULE > 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:120 > switch_loadable_module_runtime() Starting runtime thread for > mod_event_socket > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 > switch_load_network_lists() Created ip list dl-candidates default (allow) > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 10.0.0.0/8 (deny) > to list dl-candidates > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 172.16.0.0/12 > (deny) to list dl-candidates > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 192.168.0.0/16 > (deny) to list dl-candidates > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 > switch_load_network_lists() Created ip list rfc1918 default (deny) > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 10.0.0.0/8 > (allow) to list rfc1918 > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 172.16.0.0/12 > (allow) to list rfc1918 > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 192.168.0.0/16 > (allow) to list rfc1918 > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 > switch_load_network_lists() Created ip list lan default (allow) > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 192.168.42.0/24 > (deny) to list lan > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 192.168.42.42/32 > (allow) to list lan > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 > switch_load_network_lists() Created ip list strict default (deny) > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 208.102.123.124/32 > (allow) to list strict > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 > switch_load_network_lists() Created ip list domains default (deny) > 2009-04-14 12:44:26 [NOTICE] switch_core.c:965 > switch_load_network_lists() Adding 192.0.2.0/24 > (allow) [brian at 192.168.1.140 ] to list domains > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:1322 > switch_core_init_and_modload() > *FreeSWITCH Version 1.0.trunk (13013M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > > Specify what is the error why i cant able to load Mod_java. > > -- > Warm 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 From yudha2008 at gmail.com Tue Apr 14 02:37:07 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 14 Apr 2009 15:07:07 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: <49E4426C.6010400@gcdf.pl> References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> Message-ID: *Hi, I have installed latest java version jdk1.6.0 in this path /usr/java/jdk1.6.0_04/bin I have reconfigured FS ./configure --with-java=/usr/java/jdk1.6.0_04/bin, make, make install But when i run freeswitch in console i get this error. 2009-04-14 15:00:22 [ERR] modjava.c:133 load_config() Error loading /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so 2009-04-14 15:00:22 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** I cant able to load mod_java. Can any one specify what is the error. Thanks in advance. -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/7e9657e1/attachment.html From solko at gcdf.pl Tue Apr 14 03:31:24 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 14 Apr 2009 12:31:24 +0200 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> Message-ID: <49E465FC.2080305@gcdf.pl> Baskar pisze: > *Hi, > > > I have installed latest java version jdk1.6.0 in this path > /usr/java/jdk1.6.0_04/bin > > I have reconfigured FS ./configure > --with-java=/usr/java/jdk1.6.0_04/bin, make, make install > > But when i run freeswitch in console i get this error. > Exactly, configure java module in config file conf/autoload_configs/java.conf.xml This is runtime configuration not build configuration. Set right javavm path. It must point to your libjvm.so file. > 2009-04-14 15:00:22 [ERR] modjava.c:133 load_config() Error loading > /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so > 2009-04-14 15:00:22 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_java.so > **Module load routine returned an error** > > > I cant able to load mod_java. Can any one specify what is the error. > Thanks in advance. > > -- > Warm 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 From kunalgrao at yahoo.co.in Mon Apr 13 22:09:32 2009 From: kunalgrao at yahoo.co.in (kunal rao) Date: Tue, 14 Apr 2009 10:39:32 +0530 (IST) Subject: [Freeswitch-users] running on visual studio and elaborative documentation Message-ID: <542787.91121.qm@web7601.mail.in.yahoo.com> Hi ? even I have downloaded FreeSWITCH and using MS Visual Studio 2008. It is building properly. I now want to configure it properly. Can you please give me directions and also some links for good detailed comparisons between Asterisk and FreeSWITCH and elaborative documentation for the same.. --- On Wed, 1/4/09, freeswitch-users-request at lists.freeswitch.org wrote: From: freeswitch-users-request at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 34, Issue 3 To: freeswitch-users at lists.freeswitch.org Date: Wednesday, 1 April, 2009, 7:29 PM 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: Originate and Conference (Peter P GMX) ???2. Compiler error for Windows XP (SP2) (Lewis Liu) ???3. Re: Call For Help: Janitor Projects (Anthony Minessale) ???4. Re: Compiler error for Windows XP (SP2) (Michael Jerris) ???5. Re: Call For Help: Janitor Projects (Raymond Chandler) ---------------------------------------------------------------------- Message: 1 Date: Wed, 01 Apr 2009 13:41:06 +0200 From: Peter P GMX Subject: Re: [Freeswitch-users] Originate and Conference To: freeswitch-users at lists.freeswitch.org Message-ID: <49D352D2.3070303 at gmx.net> Content-Type: text/plain; charset=ISO-8859-15 Hello Brian, I tried this (on trunk 12862), but still the same behaviour. It does not aks for a PIN. Neither when transfering directly to the conference nor by transfering to the dialplan extension where conference is handled. Best regards Peter Brian West schrieb: > Update again to svn trunk... btw 1.0.4 pre3 is out on > files.freeswitch.org > > /b > > On Mar 30, 2009, at 6:44 PM, Dan Le wrote: > >> I get similar behavior as Peter when trying to enter a locked >> conference. >> >> If I am just dialing from a phone to a conference (on a dialplan), it >> will properly lock me out. But if I do an originate command >> (originate sofia/internal/1001 &conference(3000)), it will drop me >> into the conference, even though it is suppose to be locked. >> >> I am using the released 1.0.3 tag. >> > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon!? http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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: 2 Date: Wed, 1 Apr 2009 18:41:44 +0800 From: Lewis Liu Subject: [Freeswitch-users] Compiler error for Windows XP (SP2) To: freeswitch-users at lists.freeswitch.org Message-ID: ??? <814e59990904010341y61b920c2h24bb8a50c8ae2f44 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We download FreeSWITCH from SVN Trunk and want to build it on MS Visual Studio 2008 with platform. But we got one error message when we build it. FreeSWITCH\libs\win32\sofia\debug\libsofia_sip_ua_static.lib is built fail. So many files are lost, such as mod_sofia.dll..... Could you help me me for this, Please?? Whether something is lost in MS Visual Studio 2008 ?? Thanks a lot!! Lewis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/b79932de/attachment-0001.html ------------------------------ Message: 3 Date: Wed, 1 Apr 2009 08:19:54 -0500 From: Anthony Minessale Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects To: freeswitch-users at lists.freeswitch.org Message-ID: ??? <191c3a030904010619i21ddd8fj81b020340907eb27 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" have a look. http://www.google.com/search?q=janitor+project The phrase has already been coined. If you look closely we have 2 different perspectives in this thread. mszlazak is seeking more of the higher level user documentation, the holy grail magic documentation that is like the hitchhikers guide to the galaxy or harry potter's marauder's map can tune into what you need to know or what you don't understand and magically adjusts.? This is normal, we have a lot of users like that.? The majority of users will treat us like they are buying the software from us and impose their expectations on us.? It's helpful to us, it lets us see things from their perspective. Seven is looking it at more from a developer's perspective, he's actually willing to take the time to add things to the wiki and he wants to understand how the code works.? This is a good thing too, there are far less people of this type in our community but they are crucial. Core developers document by explaining what they are doing to people like Seven or by putting a reminder in the commit notes which are later translated into the CHANGELOG for the releases.? Michael, the author of this thread has added countless pages of documentation to the wiki this way. It's easy to say the author should document everything.? There is close to 300,000 lines of code in just the src directory in the FreeSWITCH tree (that is all code we wrote not counting any of the depends libs or any other form of pre-existing code).? I personally wrote the majority of that code so, I really appricate it when the communiuty gives me a few minutes to take a break while they document it.? The best people to document the high level fuctionality? is not the author btw.? It's the first few people who use it.? Most likely they are developing a product from it and they intend to profit from it in one way or another and its a fair tradeoff to have the section of functionality explained to them in exchange from wikifying it from their perspective.? The perspective of the author will be dry and mechinacal where that first-time-user version of the documentation will make much more sense to future readers. When it comes to the low level documentation, the C functions, we also need someone to help us with that if they feel there is not enough.? We write code, we know how it works.? If other people cannot figure out how it works, they will ask us and in the end it will be doucmented.? About 5% or less of people in the community even have to look in the code for the core.? The whole point of the FreeSWITCH design is to push everything up to scripts, remote connections and dialplan logic to let people concentrate on good ideas instead of the evil logic necessary to properly engineer a telephony engine.? So I recommend anybody interested starts out making sure there is ample documentation for the embedded and external API for lua, js, perl, python, ESL etc.? Then anybody who really likes C code can start with the module API layer and then dig deeper into the core code and learn how it works and if the documentation is not enough, add some, we appriciate any help we can get. 2009/4/1 >? First off. I would not call it a "janitors project" since that may offend > some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would say > you have that backwards. There is nothing more effective in evolving > FreeSwitch than good documentation which helps further development and is an > important part of "customer service." Good customer service is then a part > of "sales and marketing." Much more often than not, It's sales and marketing > that is more important to making something a "real product"? than > engineering. "Build it and they will come" almost never works. > > Anyway, I think you need a new name for this project. > > >? -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org < > freeswitch-users at lists.freeswitch.org>; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > >? Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's going to > keep growing as the word spreads. The core development team of Anthony, > Mike, and Brian are very appreciative of the community's help and > involvement in the project. Simply put: the community is awesome! > > Some have asked how they can help. Most of us are not software developers, > but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To > this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, takes > out the trash, replaces the toilet paper, etc. These are valuable services > that we sometimes take for granted. However, I think we can all appreciate > that the FreeSWITCH project would be better served if the developers could > focus on writing code, fixing bugs, etc. and not on the easier, > not-quite-as-important janitorial tasks. To that end we are inviting all who > wish to volunteer to please visit the above wiki page and check out some of > the projects listed so far. Email me off list if you'd like to volunteer to > help. I'm maintaining a list of "janitors" and what they are helping with. > If you have ideas for other janitor projects then by all means email them to > me and we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009!? http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > New Low Prices on Dell Laptops - Starting at $399 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f8bc7440/attachment-0001.html ------------------------------ Message: 4 Date: Wed, 1 Apr 2009 09:55:10 -0400 From: Michael Jerris Subject: Re: [Freeswitch-users] Compiler error for Windows XP (SP2) To: freeswitch-users at lists.freeswitch.org Message-ID: <711C4390-ED0C-4A06-9AE8-652B24D0C776 at jerris.com> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes If you try to build just the sofia library, what are the first few? warnings and errors you get? Mike On Apr 1, 2009, at 6:41 AM, Lewis Liu wrote: > We download FreeSWITCH from SVN Trunk and want to build it on MS? > Visual Studio 2008 with platform. > But we got one error message when we build it. > FreeSWITCH\libs\win32\sofia\debug\libsofia_sip_ua_static.lib is? > built fail. > So many files are lost, such as mod_sofia.dll..... > Could you help me me for this, Please?? > Whether something is lost in MS Visual Studio 2008 ?? > Thanks a lot!! > Lewis ------------------------------ Message: 5 Date: Wed, 01 Apr 2009 09:59:15 -0400 From: Raymond Chandler Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects To: freeswitch-users at lists.freeswitch.org Message-ID: <49D37333.5080701 at freeswitch.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed seven wrote: > I know that. And I'd like to read code. Developers written great code? > and also plenty of comments(which is documentation) in code. However,? > there are sth. don't need to comment in code but should be available? > on wiki. E.g. I followed the svn commit log, and found? > sip_auth_username and sip_auth_password added, so I documented to the? > wiki. >??? That's the right attitude to have... now if there were more people doing that and less people complaining like little school girls, we could actually reach the next level in Open-Sourcetopia. -Ray ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 34, Issue 3 *********************************************** Check out the all-new Messenger 9.0! Go to http://in.messenger.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/fa0dcdeb/attachment-0001.html From mike at jerris.com Tue Apr 14 04:41:01 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 14 Apr 2009 07:41:01 -0400 Subject: [Freeswitch-users] running on visual studio and elaborative documentation In-Reply-To: <542787.91121.qm@web7601.mail.in.yahoo.com> References: <542787.91121.qm@web7601.mail.in.yahoo.com> Message-ID: http://wiki.freeswitch.org/wiki/Special:Search?search=asterisk&go=Go On Apr 14, 2009, at 1:09 AM, kunal rao wrote: > > Hi > > even I have downloaded FreeSWITCH and using MS Visual Studio 2008. > It is building properly. I now want to configure it properly. Can > you please give me directions and also some links for good detailed > comparisons between Asterisk and FreeSWITCH and elaborative > documentation for the same.. > --- On Wed, 1/4/09, freeswitch-users-request at lists.freeswitch.org > wrote: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/92ed86fd/attachment.html From yudha2008 at gmail.com Tue Apr 14 04:54:04 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 14 Apr 2009 17:24:04 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: <49E465FC.2080305@gcdf.pl> References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> Message-ID: Hi, My Java.conf.xml *This is runtime configuration not build configuration. Set right javavm path. It must point to your libjvm.so file.* But still i have the same error . 2009-04-14 17:18:56 [ERR] modjava.c:133 load_config() Error loading /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so 2009-04-14 17:18:56 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** How to over come this problem. Some one help me to solve it. -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/6b5041c4/attachment.html From yudha2008 at gmail.com Tue Apr 14 05:08:15 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 14 Apr 2009 17:38:15 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> Message-ID: *Hi, I have notice While reinstalling the Freeswitch i get this message While both make and make install commands making all mod_java Note: src/org/freeswitch/Launcher.java uses unchecked or unsafe operations. Note: Recompile with -Xlint:unchecked for details. I am using CentOS 5.2 with Latest Freeswitch trunk can any one guide to resolve the problem. Thanks in advance. -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/28d78b25/attachment.html From solko at gcdf.pl Tue Apr 14 05:11:19 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 14 Apr 2009 14:11:19 +0200 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> Message-ID: <49E47D67.2080608@gcdf.pl> Baskar pisze: > Hi, > > > My Java.conf.xml > > > > > > > > > > > > *This is runtime configuration not build configuration. Set right javavm > path. > It must point to your libjvm.so file.* > > But still i have the same error . > You wrote that Java is installed in /usr/java/jdk1.6.0_04/bin so make this config file to point to it. Don't you see you are pointing wrong directory ? Find libjvm.so and put path to it in that tag. I did not gave you right configuration I just wanted to show you where to put this config value. Where is you libjvm.so located? > 2009-04-14 17:18:56 [ERR] modjava.c:133 load_config() Error loading > /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so > 2009-04-14 17:18:56 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_java.so > **Module load routine returned an error** > > How to over come this problem. Some one help me to solve it. > > -- > Warm 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 From yudha2008 at gmail.com Tue Apr 14 05:22:39 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 14 Apr 2009 17:52:39 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: <49E47D67.2080608@gcdf.pl> References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> <49E47D67.2080608@gcdf.pl> Message-ID: *Hi, I have not edited the java.conf.xml * *my libjvm.so file is loacted in this paths* * [localhost ~]# locate libjvm.so /usr/java/jdk1.6.0_04/jre/lib/i386/client/libjvm.so /usr/java/jdk1.6.0_04/jre/lib/i386/server/libjvm.so /usr/lib/gcj-4.1.1/libjvm.so /usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/i386/client/libjvm.so /usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/i386/server/libjvm.so* * In the above libjvm.so file which path should i specify it in the java.conf.xml Guide me where i am wrong. -- Warm Regards, N.Baskar* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/0972ed5f/attachment.html From solko at gcdf.pl Tue Apr 14 05:23:16 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 14 Apr 2009 14:23:16 +0200 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> Message-ID: <49E48034.1040507@gcdf.pl> Baskar pisze: > *Hi, > > > I have notice While reinstalling the Freeswitch i get this message While > both make and make install commands > > making all mod_java > Note: src/org/freeswitch/Launcher.java uses unchecked or unsafe operations. > Note: Recompile with -Xlint:unchecked for details. > > I am using CentOS 5.2 with Latest Freeswitch trunk > Don't worry about that, this is just warning. New syntax for generics in Java, but this should not lead to any problems. > can any one guide to resolve the problem. Thanks in advance. This is not your main issue. Szymon Olko > > -- > Warm 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 From anthony.minessale at gmail.com Tue Apr 14 05:57:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Apr 2009 07:57:20 -0500 Subject: [Freeswitch-users] Google Talk Integration... In-Reply-To: <29b888f80904132302k22e45594w997f70bd48be28e2@mail.gmail.com> References: <29b888f80904132302k22e45594w997f70bd48be28e2@mail.gmail.com> Message-ID: <191c3a030904140557k4709388av45288f8c7fa58248@mail.gmail.com> yes, it should be. On Tue, Apr 14, 2009 at 1:02 AM, Moiz Chinoy wrote: > Hi, > > I have tried google talk integration with FS and is working fine. Great > Work! > Is it possible to have multiple concurrent incoming calls on the same > gmail account? > > -- > Regards, > Moiz Chinoy. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/ba3d62af/attachment.html From solko at gcdf.pl Tue Apr 14 05:57:33 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 14 Apr 2009 14:57:33 +0200 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> <49E47D67.2080608@gcdf.pl> Message-ID: <49E4883D.1030305@gcdf.pl> Baskar pisze: > *Hi, > > I have not edited the java.conf.xml > * > *my libjvm.so file is loacted in this paths* > > * [localhost ~]# locate libjvm.so > /usr/java/jdk1.6.0_04/jre/lib/i386/client/libjvm.so > /usr/java/jdk1.6.0_04/jre/lib/i386/server/libjvm.so > /usr/lib/gcj-4.1.1/libjvm.so > /usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/i386/client/libjvm.so > /usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/i386/server/libjvm.so* > > * > In the above libjvm.so file which path should i specify it in the > java.conf.xml > > Guide me where i am wrong. > > Try this one first: /usr/java/jdk1.6.0_04/jre/lib/i386/client/libjvm.so if it will not work then try the one with server. This is sun Java implementation, don't sure if it will work with gcj, so don't try tem if you don't have to. > > -- > Warm 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 From anthony.minessale at gmail.com Tue Apr 14 06:15:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Apr 2009 08:15:16 -0500 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools In-Reply-To: <86a32abc0904132302y41981bd9w11f6a4c9fac5b5c3@mail.gmail.com> References: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> <86a32abc0904132253r57a4a9d3mf0f372473544276f@mail.gmail.com> <86a32abc0904132302y41981bd9w11f6a4c9fac5b5c3@mail.gmail.com> Message-ID: <191c3a030904140615p56d7f998pa5ce9a90cad62dcf@mail.gmail.com> try \d instead of \\d in your regex On Tue, Apr 14, 2009 at 1:02 AM, Diego Viola wrote: > Anthony, > > I just tried to print the variable with the log app, with read it prints, > with play_and_get_digits doesn't. > > I'm using latest SVN rev: > > FreeSWITCH Version 1.0.trunk (13012M) > > Thanks, > > Diego > > > On Tue, Apr 14, 2009 at 1:53 AM, Diego Viola wrote: > >> I remember playAndGetDigits had a bug like this too. >> >> Anthony, please help me. >> >> >> On Tue, Apr 14, 2009 at 1:42 AM, Diego Viola wrote: >> >>> It works if I use "read" and do this: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> But I need play_and_get_digits to work like that too, please. >>> >>> Diego >>> >>> >>> On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote: >>> >>>> Hi everyone, >>>> >>>> I have a question... I have this on my dialplan: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> What I want to do is play and read some digits and as soon as I get >>>> those digits, transfer to that extension... but this never happens, even if >>>> I terminate with a #. >>>> >>>> I do the same thing with Lua and it works with Lua, but I need it to >>>> work with play_and_get_digits from mod_dptools, because I plan to use this >>>> with event socket outbound, with an application which I'm currently working >>>> on. >>>> >>>> Any ideas? >>>> >>>> Thanks, >>>> >>>> Diego >>>> >>> >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/fc0876f3/attachment.html From peter.olsson at visionutveckling.se Tue Apr 14 06:24:26 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 14 Apr 2009 15:24:26 +0200 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> Anthony, Yes, it seems to work correct now. I did a couple of test calls, and tha audio was good - thanks! Another question about this scenario... When doing a session.transfer("5000"), this will transfer the call directly into the dialplan without the use of loopback-channels. But that way it's not possible to do it in a controlled way. Shouldn't it be possible to do the same thing with a bridge? As soon as the call is bridged, it gets "rid of" unneccecary loopback channels, and connecting the two endpoints directly - cause by then it should be two "normal" endpoints talking? Regards, Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 13 april 2009 20:38 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... see how it works in latest trunk 13011 nontheless you can just say session.execute("bridge", "loopback/5000"); and get the same result without touching that other channel. when the call fails, you will have an originate_disposition variable in session you can check. On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson > wrote: 1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available.. 2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg. 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. :) What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected? --- if (session.ready()) { session.answer(); new_session = new Session("loopback/5000", session); new_session.waitForAnswer(); bridge(session, new_session); // Not sure if this is needed - I've tried with it both enabled and disabled session.hangup(); new_session.hangup(); } Peter On 09-04-13 17.54, "Anthony Minessale" > wrote: 1) When you say latest, which rev does that mean? we change revs pretty often. 2) Do you have a minimal script that reproduces your issue. 3) is there a reason you cannot just session.execute("bridge", dest); instead of doing it manually (which is a process not for the faint at heart)? On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson > wrote: I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 !DSPAM:49e3899632939315582408! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/c290958f/attachment-0001.html From gmaruzz at celliax.org Tue Apr 14 07:13:26 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 14 Apr 2009 16:13:26 +0200 Subject: [Freeswitch-users] skypiax Round Robin interface In-Reply-To: References: Message-ID: <7b197bef0904140713w69d7916au8f319ac37c138c11@mail.gmail.com> Hi Seven, thanks a lot for the patch and all the Skypiax action. I'm just back from Eastern vacations, let me clear the backlog and I'll be back on this in a couple days. Thanks again! gm Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Fri, Apr 10, 2009 at 8:38 PM, dujinfang wrote: > Hi, > > I made a patch, so skypiax is possible to do a RR hunt besides the > sequential interface ANY. > > Usage: > > originate skypiax/RR/other_skype_name > sk list > > http://jira.freeswitch.org/browse/MODENDP-211 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 gmaruzz at celliax.org Tue Apr 14 07:27:44 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 14 Apr 2009 16:27:44 +0200 Subject: [Freeswitch-users] Skypiax as a windows service In-Reply-To: <0D1E9E22CCAC4F98ADA9863FDFF7FB85@UVix> References: <0D1E9E22CCAC4F98ADA9863FDFF7FB85@UVix> Message-ID: <7b197bef0904140727o693e1e72jbf7470cc00afbb9d@mail.gmail.com> Hi UV, seems a difficult one this one. I have no much experience in RDP/terminal server. If there is no way to have (or fake) audio driver on RDP/terminal server apps, probably the Skype clients will not works (as you experienced). I'm sure, I've read it (:-) ), that Skype clients can be run on a Windows machine as services, without any user logged in. That is what I would explore in the future, just adding the How To to the wiki page. What you are experiencing seems to be different, seems to be specific to the RDP/terminal server usage. I'm I understanding you correctly (that this is specific to RDP)? Can you send me more info/hints? In parallel, I'm slowly working on a way to farm out the Skype clients from the FS servers, so to have the Skype clients running on different machines on the same LAN. I've a proof of concept working on Linux for one channel. You think this would solve your problems (having the Skype clients running on separate machines other than the machines running FS)? I'm just back from Easter vacations, please allow a couple days for the accumulated backlog ;-) Thanks a lot for taking the time to explore Skypiax and report this, gm Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Mon, Apr 13, 2009 at 1:32 PM, UV wrote: > Great work on Skypiax, Giovanni. > > > > We?ve tested it in our lab for sometime and it works very well. > > Unfortunately, when we tried deploying it on a production environment > (running Win2K3 server farm), we ran into a barrier: > > FS is running as terminal server console application (to be easily > maintained remotely by RDP) > This is because Win2K does not allow RDP to access system console (session > /userid 0) > Skype does not work on terminal server due to a well known disappearing > audio drivers problem, therefore it has to run either as a console or a > service (both on session 0). > FS can run well as a windows service > Skypiax seem to load as service, but it can?t find the skype client and exit > with the following error: > > 2009-04-13 20:54:14 [ERR] mod_skypiax.c:990 load_config() rev > 13006M[00000000|37???? ][ERRORA? 990? ][skype_user??? ][-1, 0, 0] Failed to > connect to a SKYPE API for interface_id=1, no SKYPE client running, please > (re)start Skype client. Skypiax exiting > > > > This situation prevents me to run skypiax in production. > > > > I understand from the wiki page that windows service is not done yet ? so I > presume this is a predicted outcome. > > > > Any idea when and if this is planned to be implemented? > > > > Keep up the good work! > > > > Cheers, > > UV > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Tue Apr 14 08:26:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Apr 2009 10:26:55 -0500 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> Message-ID: <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> yes, But if you plan is to bridge the call, the loopback channel is completely unnecessary. Be careful how much control you want =D getting a phone call up and running is more work than you think (see switch_ivr_originate.c) On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Anthony, > > > > Yes, it seems to work correct now. I did a couple of test calls, and tha > audio was good ? thanks! > > > > Another question about this scenario... > > > > When doing a session.transfer(?5000?), this will transfer the call directly > into the dialplan without the use of loopback-channels. But that way it?s > not possible to do it in a controlled way. Shouldn?t it be possible to do > the same thing with a bridge? As soon as the call is bridged, it gets ?rid > of? unneccecary loopback channels, and connecting the two endpoints directly > ? cause by then it should be two ?normal? endpoints talking? > > > > Regards, > > > > Peter > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 13 april 2009 20:38 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > see how it works in latest trunk 13011 > > nontheless you can just say > > session.execute("bridge", "loopback/5000"); > > and get the same result without touching that other channel. > > when the call fails, you will have an originate_disposition variable in > session you can check. > > > On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > 1. The latest trunk I've tried with is 13008. Since I'm not doing > anything for production yet (just testing/evaluating), so I tend to update > as soon as there is new version available.. > 2. Yep, you will find it below. In javascript - my sample for .NET does > basically the same thing, with the same result, except that it also won't > drop the loopback-a call leg. > 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess > I'm not 100% sure what I'm doing.. :) What I want to be able to do is to > dial into a script, let the script dial another extension, and bridge them > together when the other party answers the call. I also need to take care of > call setup problems - if the other part doesn't respond, is unavailable or > busy in the phone - so I though this was the only way? If I use the > session.execute("bridge"..), will I be able to control the call if it > couldn't be connected? > > --- > > if (session.ready()) { > > session.answer(); > > new_session = new Session("loopback/5000", session); > new_session.waitForAnswer(); > > bridge(session, new_session); > > // Not sure if this is needed - I've tried with it both enabled and > disabled > session.hangup(); > new_session.hangup(); > } > > Peter > > > > On 09-04-13 17.54, "Anthony Minessale" > wrote: > > 1) When you say latest, which rev does that mean? we change revs pretty > often. > 2) Do you have a minimal script that reproduces your issue. > 3) is there a reason you cannot just session.execute("bridge", dest); > instead of doing it manually (which is a process not for the faint at > heart)? > > > > On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > I have two problems that I haven't been able to solve. I've done the same > tests in both javascript, and in .NET. > > The two scripts are pretty simple, they just answer an incomming call, > creates a new session, wait for an answer on the second call leg, and then > bridge the two channels together. > > In both cases everything works just fine, but the audio is distorted. The > destination I'm calling is "loopback/5000" - the sample IVR application > included in FreeSWITCH. I first thought it was a codec issue, but even after > trying to switch to different codecs the problem was the same. It more > sounds like it's a timestamping issue - the voice is not distorted enough to > be a bad codec, but it reads way to fast (mayby twice the "normal" speed). > When doing a direct transfer() to the other destination this works just > fine, but I need to be able to have some extra logic to tell if the > destination is available or not. > > The second problem occurs only in .NET. After doing this sample there is as > loopback channel still hanging around. It seems like the call creates a > loopback-a and loopback-b, the loopback-b dissapears as it should (when the > call has been disconnected), but the other one stays there. When doing the > same in javascript this doesn't seem to occur. > > I'm using the latest SVN trunk, and my OS is Windows XP. > > I found bug FSCORE-349 in Jira, which seems to point in to the direction > that there might be a bug with the loopback channels in some cases, but I > could not find anything about the audio which plays too fast. > > Has anyone else experienced this? > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > !DSPAM:49e3899632939315582408! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/e38e0519/attachment.html From peter.olsson at visionutveckling.se Tue Apr 14 08:59:13 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 14 Apr 2009 17:59:13 +0200 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE413034@cooper> Yes, I'm starting to realize that... :) but you to get everything right - if I want to bridge a call, using the dialplan, then the only way is to use loopback, right? If I don't want a loopback I'm able to bridge to the destination directly? //Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 14 april 2009 17:27 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... yes, But if you plan is to bridge the call, the loopback channel is completely unnecessary. Be careful how much control you want =D getting a phone call up and running is more work than you think (see switch_ivr_originate.c) On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson > wrote: Anthony, Yes, it seems to work correct now. I did a couple of test calls, and tha audio was good - thanks! Another question about this scenario... When doing a session.transfer("5000"), this will transfer the call directly into the dialplan without the use of loopback-channels. But that way it's not possible to do it in a controlled way. Shouldn't it be possible to do the same thing with a bridge? As soon as the call is bridged, it gets "rid of" unneccecary loopback channels, and connecting the two endpoints directly - cause by then it should be two "normal" endpoints talking? Regards, Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 13 april 2009 20:38 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... see how it works in latest trunk 13011 nontheless you can just say session.execute("bridge", "loopback/5000"); and get the same result without touching that other channel. when the call fails, you will have an originate_disposition variable in session you can check. On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson > wrote: 1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available.. 2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg. 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. :) What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected? --- if (session.ready()) { session.answer(); new_session = new Session("loopback/5000", session); new_session.waitForAnswer(); bridge(session, new_session); // Not sure if this is needed - I've tried with it both enabled and disabled session.hangup(); new_session.hangup(); } Peter On 09-04-13 17.54, "Anthony Minessale" > wrote: 1) When you say latest, which rev does that mean? we change revs pretty often. 2) Do you have a minimal script that reproduces your issue. 3) is there a reason you cannot just session.execute("bridge", dest); instead of doing it manually (which is a process not for the faint at heart)? On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson > wrote: I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 !DSPAM:49e4ade432931915915389! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/f882100e/attachment-0001.html From anthony.minessale at gmail.com Tue Apr 14 09:25:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Apr 2009 11:25:37 -0500 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE413034@cooper> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE413034@cooper> Message-ID: <191c3a030904140925x28e322bft389091547ca0d5d6@mail.gmail.com> The bridge application will let you bridge right to a destination on *another* box. If you want to connect to a local extension like 5000 you can use the transfer application or method. session.transfer("5000"); exit(); or session.execute("transfer", "5000"); exit(); On Tue, Apr 14, 2009 at 10:59 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Yes, I?m starting to realize that... :) but you to get everything right ? > if I want to bridge a call, using the dialplan, then the only way is to use > loopback, right? If I don?t want a loopback I?m able to bridge to the > destination directly? > > > > //Peter > > > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 14 april 2009 17:27 > > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > yes, > > But if you plan is to bridge the call, the loopback channel is completely > unnecessary. > Be careful how much control you want =D getting a phone call up and running > is more work > than you think (see switch_ivr_originate.c) > > On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > Anthony, > > > > Yes, it seems to work correct now. I did a couple of test calls, and tha > audio was good ? thanks! > > > > Another question about this scenario... > > > > When doing a session.transfer(?5000?), this will transfer the call directly > into the dialplan without the use of loopback-channels. But that way it?s > not possible to do it in a controlled way. Shouldn?t it be possible to do > the same thing with a bridge? As soon as the call is bridged, it gets ?rid > of? unneccecary loopback channels, and connecting the two endpoints directly > ? cause by then it should be two ?normal? endpoints talking? > > > > Regards, > > > > Peter > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 13 april 2009 20:38 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > see how it works in latest trunk 13011 > > nontheless you can just say > > session.execute("bridge", "loopback/5000"); > > and get the same result without touching that other channel. > > when the call fails, you will have an originate_disposition variable in > session you can check. > > On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > 1. The latest trunk I've tried with is 13008. Since I'm not doing > anything for production yet (just testing/evaluating), so I tend to update > as soon as there is new version available.. > 2. Yep, you will find it below. In javascript - my sample for .NET does > basically the same thing, with the same result, except that it also won't > drop the loopback-a call leg. > 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess > I'm not 100% sure what I'm doing.. :) What I want to be able to do is to > dial into a script, let the script dial another extension, and bridge them > together when the other party answers the call. I also need to take care of > call setup problems - if the other part doesn't respond, is unavailable or > busy in the phone - so I though this was the only way? If I use the > session.execute("bridge"..), will I be able to control the call if it > couldn't be connected? > > --- > > if (session.ready()) { > > session.answer(); > > new_session = new Session("loopback/5000", session); > new_session.waitForAnswer(); > > bridge(session, new_session); > > // Not sure if this is needed - I've tried with it both enabled and > disabled > session.hangup(); > new_session.hangup(); > } > > Peter > > > > On 09-04-13 17.54, "Anthony Minessale" > wrote: > > 1) When you say latest, which rev does that mean? we change revs pretty > often. > 2) Do you have a minimal script that reproduces your issue. > 3) is there a reason you cannot just session.execute("bridge", dest); > instead of doing it manually (which is a process not for the faint at > heart)? > > > > On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > I have two problems that I haven't been able to solve. I've done the same > tests in both javascript, and in .NET. > > The two scripts are pretty simple, they just answer an incomming call, > creates a new session, wait for an answer on the second call leg, and then > bridge the two channels together. > > In both cases everything works just fine, but the audio is distorted. The > destination I'm calling is "loopback/5000" - the sample IVR application > included in FreeSWITCH. I first thought it was a codec issue, but even after > trying to switch to different codecs the problem was the same. It more > sounds like it's a timestamping issue - the voice is not distorted enough to > be a bad codec, but it reads way to fast (mayby twice the "normal" speed). > When doing a direct transfer() to the other destination this works just > fine, but I need to be able to have some extra logic to tell if the > destination is available or not. > > The second problem occurs only in .NET. After doing this sample there is as > loopback channel still hanging around. It seems like the call creates a > loopback-a and loopback-b, the loopback-b dissapears as it should (when the > call has been disconnected), but the other one stays there. When doing the > same in javascript this doesn't seem to occur. > > I'm using the latest SVN trunk, and my OS is Windows XP. > > I found bug FSCORE-349 in Jira, which seems to point in to the direction > that there might be a bug with the loopback channels in some cases, but I > could not find anything about the audio which plays too fast. > > Has anyone else experienced this? > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > !DSPAM:49e4ade432931915915389! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/e0ce86cd/attachment.html From kristian.kielhofner at gmail.com Tue Apr 14 09:51:37 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 14 Apr 2009 12:51:37 -0400 Subject: [Freeswitch-users] Adding Spanish support to say Message-ID: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> Hello everyone, I'm trying to add Spanish support to say. I'm using something like: in conf/lang/es which is included by freeswitch.conf: ..right after English. Yet I continue to get [ERR] switch_ivr.c:2014 switch_ivr_say() Invalid SAY Interface [es]! Whenever trying to use say: What am I missing? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Tue Apr 14 09:54:54 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 14 Apr 2009 12:54:54 -0400 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> Message-ID: <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> Replying to myself... I forgot to indicate my version! I am running trunk rev 12862 on CentOS 5 x86_64. On Tue, Apr 14, 2009 at 12:51 PM, Kristian Kielhofner wrote: > Hello everyone, > > ?I'm trying to add Spanish support to say. ?I'm using something like: > > > ? tts-engine="cepstral" tts-voice="callie"> > ? ? > ? ? > ? ? ? > ? > > > in conf/lang/es which is included by freeswitch.conf: > > > > ..right after English. ?Yet I continue to get > > [ERR] switch_ivr.c:2014 switch_ivr_say() Invalid SAY Interface [es]! > > ?Whenever trying to use say: > > > > ?What am I missing? > > Thanks! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Tue Apr 14 10:01:22 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Apr 2009 12:01:22 -0500 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> Message-ID: Nobody has written the es language files. Those would need to be written. /b On Apr 14, 2009, at 11:54 AM, Kristian Kielhofner wrote: > Replying to myself... I forgot to indicate my version! I am running > trunk rev 12862 on CentOS 5 x86_64. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/0a366926/attachment.html From peter.olsson at visionutveckling.se Tue Apr 14 10:02:47 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 14 Apr 2009 19:02:47 +0200 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <191c3a030904140925x28e322bft389091547ca0d5d6@mail.gmail.com> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE413034@cooper> <191c3a030904140925x28e322bft389091547ca0d5d6@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE41303C@cooper> Allright - last question :) I'll try to be a little more specific. Lets say I whant to do the following; 1. Dial into FreeSWITCH, to some kind of application (javascript or whatever). 2. Answer that call, and let the user choose what to do; 1: record message, 2: transfer to XXX etc. The user presses 2. 3. I don't want to release the first call leg yet, since I need to be really sure that 2 is reachable (or else I will give the user choices again, with som kind of "the call could not be transferred"). So lets say I play some music for the user while trying to connect the call. 4. I originate another call - now I understand I have two choices, either I originate directly to a SIP phone (sofia/internal...), or I let the dialplan do the work - and if I want the dialplan to be the one to transfer the call somewhere (maybe to the same extension), I must use loopback - right? 5. If the new call answers, bridge the two calls, if it fails, start over again, after reading an error message. Whould this also be possible with transfer? If I understand everything right I loose control of the call, and won't be able to handle the failed transfer? Or is it possible to solve in a better way? What I guess I'd really want to do is to ask the dialplan "hey, I want to dial XXXX - give me the full sofia profile string" so I can originate the call directly, and I won't need a loopback. I could of course connect to the sofia string directly, but it would be nice to leave that kind of lookup logic to the dialplan. Thanks for staying with me - I hope you understand my problem :) Regards, Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 14 april 2009 18:26 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... The bridge application will let you bridge right to a destination on *another* box. If you want to connect to a local extension like 5000 you can use the transfer application or method. session.transfer("5000"); exit(); or session.execute("transfer", "5000"); exit(); On Tue, Apr 14, 2009 at 10:59 AM, Peter Olsson > wrote: Yes, I'm starting to realize that... :) but you to get everything right - if I want to bridge a call, using the dialplan, then the only way is to use loopback, right? If I don't want a loopback I'm able to bridge to the destination directly? //Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 14 april 2009 17:27 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... yes, But if you plan is to bridge the call, the loopback channel is completely unnecessary. Be careful how much control you want =D getting a phone call up and running is more work than you think (see switch_ivr_originate.c) On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson > wrote: Anthony, Yes, it seems to work correct now. I did a couple of test calls, and tha audio was good - thanks! Another question about this scenario... When doing a session.transfer("5000"), this will transfer the call directly into the dialplan without the use of loopback-channels. But that way it's not possible to do it in a controlled way. Shouldn't it be possible to do the same thing with a bridge? As soon as the call is bridged, it gets "rid of" unneccecary loopback channels, and connecting the two endpoints directly - cause by then it should be two "normal" endpoints talking? Regards, Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 13 april 2009 20:38 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... see how it works in latest trunk 13011 nontheless you can just say session.execute("bridge", "loopback/5000"); and get the same result without touching that other channel. when the call fails, you will have an originate_disposition variable in session you can check. On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson > wrote: 1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available.. 2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg. 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. :) What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected? --- if (session.ready()) { session.answer(); new_session = new Session("loopback/5000", session); new_session.waitForAnswer(); bridge(session, new_session); // Not sure if this is needed - I've tried with it both enabled and disabled session.hangup(); new_session.hangup(); } Peter On 09-04-13 17.54, "Anthony Minessale" > wrote: 1) When you say latest, which rev does that mean? we change revs pretty often. 2) Do you have a minimal script that reproduces your issue. 3) is there a reason you cannot just session.execute("bridge", dest); instead of doing it manually (which is a process not for the faint at heart)? On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson > wrote: I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 !DSPAM:49e4bcb132932104520616! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/c1d31444/attachment-0001.html From anthony.minessale at gmail.com Tue Apr 14 10:21:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Apr 2009 12:21:19 -0500 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE41303C@cooper> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE413034@cooper> <191c3a030904140925x28e322bft389091547ca0d5d6@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE41303C@cooper> Message-ID: <191c3a030904141021mf4d873bvf361135dd677ea0f@mail.gmail.com> typically you would use transfer to the dest then in the dialplan for XXXX you would set hangup_after_bridge=true try to call the phone transfer back to your ivr you can use channel variables to keep track of state. On Tue, Apr 14, 2009 at 12:02 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Allright ? last question :) I?ll try to be a little more specific. Lets > say I whant to do the following; > > > > 1. Dial into FreeSWITCH, to some kind of application (javascript or > whatever). > > 2. Answer that call, and let the user choose what to do; 1: record > message, 2: transfer to XXX etc. The user presses 2. > > 3. I don?t want to release the first call leg yet, since I need to > be really sure that 2 is reachable (or else I will give the user choices > again, with som kind of ?the call could not be transferred?). So lets say I > play some music for the user while trying to connect the call. > > 4. I originate another call ? now I understand I have two choices, > either I originate directly to a SIP phone (sofia/internal...), or I let the > dialplan do the work ? and if I want the dialplan to be the one to transfer > the call somewhere (maybe to the same extension), I must use loopback ? > right? > > 5. If the new call answers, bridge the two calls, if it fails, start > over again, after reading an error message. > > > > Whould this also be possible with transfer? If I understand everything > right I loose control of the call, and won?t be able to handle the failed > transfer? Or is it possible to solve in a better way? > > > > What I guess I?d really want to do is to ask the dialplan ?hey, I want to > dial XXXX ? give me the full sofia profile string? so I can originate the > call directly, and I won?t need a loopback. I could of course connect to the > sofia string directly, but it would be nice to leave that kind of lookup > logic to the dialplan. > > > > Thanks for staying with me ? I hope you understand my problem :) > > > > Regards, > > > > Peter > > > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 14 april 2009 18:26 > > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > The bridge application will let you bridge right to a destination on > *another* box. > If you want to connect to a local extension like 5000 you can use the > transfer application or method. > > session.transfer("5000"); > exit(); > > or > > session.execute("transfer", "5000"); > exit(); > > > On Tue, Apr 14, 2009 at 10:59 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > Yes, I?m starting to realize that... :) but you to get everything right ? > if I want to bridge a call, using the dialplan, then the only way is to use > loopback, right? If I don?t want a loopback I?m able to bridge to the > destination directly? > > > > //Peter > > > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 14 april 2009 17:27 > > > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > yes, > > But if you plan is to bridge the call, the loopback channel is completely > unnecessary. > Be careful how much control you want =D getting a phone call up and running > is more work > than you think (see switch_ivr_originate.c) > > On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > Anthony, > > > > Yes, it seems to work correct now. I did a couple of test calls, and tha > audio was good ? thanks! > > > > Another question about this scenario... > > > > When doing a session.transfer(?5000?), this will transfer the call directly > into the dialplan without the use of loopback-channels. But that way it?s > not possible to do it in a controlled way. Shouldn?t it be possible to do > the same thing with a bridge? As soon as the call is bridged, it gets ?rid > of? unneccecary loopback channels, and connecting the two endpoints directly > ? cause by then it should be two ?normal? endpoints talking? > > > > Regards, > > > > Peter > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 13 april 2009 20:38 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > see how it works in latest trunk 13011 > > nontheless you can just say > > session.execute("bridge", "loopback/5000"); > > and get the same result without touching that other channel. > > when the call fails, you will have an originate_disposition variable in > session you can check. > > On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > 1. The latest trunk I've tried with is 13008. Since I'm not doing > anything for production yet (just testing/evaluating), so I tend to update > as soon as there is new version available.. > 2. Yep, you will find it below. In javascript - my sample for .NET does > basically the same thing, with the same result, except that it also won't > drop the loopback-a call leg. > 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess > I'm not 100% sure what I'm doing.. :) What I want to be able to do is to > dial into a script, let the script dial another extension, and bridge them > together when the other party answers the call. I also need to take care of > call setup problems - if the other part doesn't respond, is unavailable or > busy in the phone - so I though this was the only way? If I use the > session.execute("bridge"..), will I be able to control the call if it > couldn't be connected? > > --- > > if (session.ready()) { > > session.answer(); > > new_session = new Session("loopback/5000", session); > new_session.waitForAnswer(); > > bridge(session, new_session); > > // Not sure if this is needed - I've tried with it both enabled and > disabled > session.hangup(); > new_session.hangup(); > } > > Peter > > > > On 09-04-13 17.54, "Anthony Minessale" > wrote: > > 1) When you say latest, which rev does that mean? we change revs pretty > often. > 2) Do you have a minimal script that reproduces your issue. > 3) is there a reason you cannot just session.execute("bridge", dest); > instead of doing it manually (which is a process not for the faint at > heart)? > > > > On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > I have two problems that I haven't been able to solve. I've done the same > tests in both javascript, and in .NET. > > The two scripts are pretty simple, they just answer an incomming call, > creates a new session, wait for an answer on the second call leg, and then > bridge the two channels together. > > In both cases everything works just fine, but the audio is distorted. The > destination I'm calling is "loopback/5000" - the sample IVR application > included in FreeSWITCH. I first thought it was a codec issue, but even after > trying to switch to different codecs the problem was the same. It more > sounds like it's a timestamping issue - the voice is not distorted enough to > be a bad codec, but it reads way to fast (mayby twice the "normal" speed). > When doing a direct transfer() to the other destination this works just > fine, but I need to be able to have some extra logic to tell if the > destination is available or not. > > The second problem occurs only in .NET. After doing this sample there is as > loopback channel still hanging around. It seems like the call creates a > loopback-a and loopback-b, the loopback-b dissapears as it should (when the > call has been disconnected), but the other one stays there. When doing the > same in javascript this doesn't seem to occur. > > I'm using the latest SVN trunk, and my OS is Windows XP. > > I found bug FSCORE-349 in Jira, which seems to point in to the direction > that there might be a bug with the loopback channels in some cases, but I > could not find anything about the audio which plays too fast. > > Has anyone else experienced this? > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > !DSPAM:49e4bcb132932104520616! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/107b32fe/attachment-0001.html From msc at freeswitch.org Tue Apr 14 10:25:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Apr 2009 10:25:34 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre4 Now Available Message-ID: <87f2f3b90904141025h4df3320ai60d9710ab6449f26@mail.gmail.com> The FreeSWITCH team is pleased to announce the immediate availability of version 1.0.4pre4. Details are available here: http://www.freeswitch.org/node/173 All are encouraged to upgrade as soon as possible. Thanks to everyone for their feedback, ideas, and bug reports. Please keep them coming. -Michael http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/16f9e5b0/attachment.html From msc at freeswitch.org Tue Apr 14 10:37:27 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Apr 2009 10:37:27 -0700 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> Message-ID: <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> KK, Do you have someone who knows Spanish and who can translate? If not I will whip up some volunteers from the FS community. Thanks, MC On Tue, Apr 14, 2009 at 10:01 AM, Brian West wrote: > Nobody has written the es language files. Those would need to be written. > /b > > On Apr 14, 2009, at 11:54 AM, Kristian Kielhofner wrote: > > Replying to myself... I forgot to indicate my version! I am running > trunk rev 12862 on CentOS 5 x86_64. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/b4a2497f/attachment.html From jmesquita at gmail.com Tue Apr 14 10:48:38 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 14 Apr 2009 14:48:38 -0300 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> Message-ID: <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> I know spanish and I would translate it no problem. MC, get in touch with me off-list so we can handle that. I can also translate to portuguese-brazil. jmesquita On Apr 14, 2009, at 2:37 PM, Michael Collins wrote: > KK, > Do you have someone who knows Spanish and who can translate? If not > I will whip up some volunteers from the FS community. > > Thanks, > MC > > On Tue, Apr 14, 2009 at 10:01 AM, Brian West > wrote: > Nobody has written the es language files. Those would need to be > written. > > /b > > On Apr 14, 2009, at 11:54 AM, Kristian Kielhofner wrote: > >> Replying to myself... I forgot to indicate my version! I am running >> trunk rev 12862 on CentOS 5 x86_64. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/990f33ae/attachment.html From brian at freeswitch.org Tue Apr 14 10:57:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Apr 2009 12:57:44 -0500 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> Message-ID: <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> This also requires you to write all the phrase macros for voicemail, ivr and other things in the demo in lang/en/ /b On Apr 14, 2009, at 12:48 PM, Jo?o Mesquita wrote: > I know spanish and I would translate it no problem. MC, get in touch > with me off-list so we can handle that. > > I can also translate to portuguese-brazil. > > jmesquita Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/9979ea58/attachment.html From kristian.kielhofner at gmail.com Tue Apr 14 11:07:14 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 14 Apr 2009 14:07:14 -0400 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> Message-ID: <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> Brian, For my application I just need to be able to say a string of numbers - Caller ID, etc. Other than the files used there is no syntax or grammar difference (in Spanish) when compared to English. I should just be able to drop the files in. I'll have a problem when I need to handle IVR, voicemail, and other more complex issues but this will solve my immediate needs. For now I'm just trying to figure out how to get language "es" recognized by say... On Tue, Apr 14, 2009 at 1:57 PM, Brian West wrote: > This also requires you to write all the phrase macros for voicemail, ivr and > other things in the demo in lang/en/ > /b > On Apr 14, 2009, at 12:48 PM, Jo?o Mesquita wrote: > > I know spanish and I would translate it no problem. MC, get in touch with me > off-list so we can handle that. > I can also translate to portuguese-brazil. > jmesquita > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From msc at freeswitch.org Tue Apr 14 11:12:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Apr 2009 11:12:20 -0700 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> Message-ID: <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> Cool. We've had several volunteers start translating the phrase files into Spanish and Brazilian Portugese. We'll keep you posted when we have the Spanish one ready. FYI, I committed a stub phrase_es.xml file but it hasn't been translated yet except for the first twenty digits. However, there aren't any audio files associated with it yet... -MC On Tue, Apr 14, 2009 at 11:07 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Brian, > > For my application I just need to be able to say a string of numbers > - Caller ID, etc. > > Other than the files used there is no syntax or grammar difference > (in Spanish) when compared to English. I should just be able to drop > the files in. > > I'll have a problem when I need to handle IVR, voicemail, and other > more complex issues but this will solve my immediate needs. > > For now I'm just trying to figure out how to get language "es" > recognized by say... > > On Tue, Apr 14, 2009 at 1:57 PM, Brian West wrote: > > This also requires you to write all the phrase macros for voicemail, ivr > and > > other things in the demo in lang/en/ > > /b > > On Apr 14, 2009, at 12:48 PM, Jo?o Mesquita wrote: > > > > I know spanish and I would translate it no problem. MC, get in touch with > me > > off-list so we can handle that. > > I can also translate to portuguese-brazil. > > jmesquita > > > > Brian West > > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/553c3d06/attachment-0001.html From diego.viola at gmail.com Tue Apr 14 11:48:40 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 14:48:40 -0400 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools In-Reply-To: <191c3a030904140615p56d7f998pa5ce9a90cad62dcf@mail.gmail.com> References: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> <86a32abc0904132253r57a4a9d3mf0f372473544276f@mail.gmail.com> <86a32abc0904132302y41981bd9w11f6a4c9fac5b5c3@mail.gmail.com> <191c3a030904140615p56d7f998pa5ce9a90cad62dcf@mail.gmail.com> Message-ID: <86a32abc0904141148l33454b54m533cb556bd8f6515@mail.gmail.com> That works, thanks Anthm, you're the man. Diego On Tue, Apr 14, 2009 at 9:15 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try \d instead of \\d in your regex > > On Tue, Apr 14, 2009 at 1:02 AM, Diego Viola wrote: > >> Anthony, >> >> I just tried to print the variable with the log app, with read it prints, >> with play_and_get_digits doesn't. >> >> I'm using latest SVN rev: >> >> FreeSWITCH Version 1.0.trunk (13012M) >> >> Thanks, >> >> Diego >> >> >> On Tue, Apr 14, 2009 at 1:53 AM, Diego Viola wrote: >> >>> I remember playAndGetDigits had a bug like this too. >>> >>> Anthony, please help me. >>> >>> >>> On Tue, Apr 14, 2009 at 1:42 AM, Diego Viola wrote: >>> >>>> It works if I use "read" and do this: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> But I need play_and_get_digits to work like that too, please. >>>> >>>> Diego >>>> >>>> >>>> On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote: >>>> >>>>> Hi everyone, >>>>> >>>>> I have a question... I have this on my dialplan: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> What I want to do is play and read some digits and as soon as I get >>>>> those digits, transfer to that extension... but this never happens, even if >>>>> I terminate with a #. >>>>> >>>>> I do the same thing with Lua and it works with Lua, but I need it to >>>>> work with play_and_get_digits from mod_dptools, because I plan to use this >>>>> with event socket outbound, with an application which I'm currently working >>>>> on. >>>>> >>>>> Any ideas? >>>>> >>>>> Thanks, >>>>> >>>>> Diego >>>>> >>>> >>>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/73eed9e9/attachment.html From diego.viola at gmail.com Tue Apr 14 11:56:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 14:56:02 -0400 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> Message-ID: <86a32abc0904141156x4bd04908pa0a83854d4617f56@mail.gmail.com> Hey guys, If you need some Spanish help count with my help also. Diego On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins wrote: > Cool. We've had several volunteers start translating the phrase files into > Spanish and Brazilian Portugese. We'll keep you posted when we have the > Spanish one ready. FYI, I committed a stub phrase_es.xml file but it hasn't > been translated yet except for the first twenty digits. However, there > aren't any audio files associated with it yet... > > -MC > > > On Tue, Apr 14, 2009 at 11:07 AM, Kristian Kielhofner < > kristian.kielhofner at gmail.com> wrote: > >> Brian, >> >> For my application I just need to be able to say a string of numbers >> - Caller ID, etc. >> >> Other than the files used there is no syntax or grammar difference >> (in Spanish) when compared to English. I should just be able to drop >> the files in. >> >> I'll have a problem when I need to handle IVR, voicemail, and other >> more complex issues but this will solve my immediate needs. >> >> For now I'm just trying to figure out how to get language "es" >> recognized by say... >> >> On Tue, Apr 14, 2009 at 1:57 PM, Brian West wrote: >> > This also requires you to write all the phrase macros for voicemail, ivr >> and >> > other things in the demo in lang/en/ >> > /b >> > On Apr 14, 2009, at 12:48 PM, Jo?o Mesquita wrote: >> > >> > I know spanish and I would translate it no problem. MC, get in touch >> with me >> > off-list so we can handle that. >> > I can also translate to portuguese-brazil. >> > jmesquita >> > >> > Brian West >> > brian at freeswitch.org >> > -- Meet us at ClueCon! http://www.cluecon.com >> > >> > >> > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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://blog.krisk.org >> http://www.submityoursip.com >> http://www.astlinux.org >> http://www.star2star.com >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/30398a21/attachment.html From nicolas at medularis.com Tue Apr 14 12:21:02 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 14 Apr 2009 15:21:02 -0400 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <86a32abc0904141156x4bd04908pa0a83854d4617f56@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> <86a32abc0904141156x4bd04908pa0a83854d4617f56@mail.gmail.com> Message-ID: <1b46b4e80904141221h16cfa78ci807f8541f66b31b1@mail.gmail.com> I'm a native spanish speaker, I can help too! Nicol?s Brenner On Tue, Apr 14, 2009 at 2:56 PM, Diego Viola wrote: > Hey guys, > > If you need some Spanish help count with my help also. > > Diego > > > On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins wrote: > >> Cool. We've had several volunteers start translating the phrase files into >> Spanish and Brazilian Portugese. We'll keep you posted when we have the >> Spanish one ready. FYI, I committed a stub phrase_es.xml file but it hasn't >> been translated yet except for the first twenty digits. However, there >> aren't any audio files associated with it yet... >> >> -MC >> >> >> On Tue, Apr 14, 2009 at 11:07 AM, Kristian Kielhofner < >> kristian.kielhofner at gmail.com> wrote: >> >>> Brian, >>> >>> For my application I just need to be able to say a string of numbers >>> - Caller ID, etc. >>> >>> Other than the files used there is no syntax or grammar difference >>> (in Spanish) when compared to English. I should just be able to drop >>> the files in. >>> >>> I'll have a problem when I need to handle IVR, voicemail, and other >>> more complex issues but this will solve my immediate needs. >>> >>> For now I'm just trying to figure out how to get language "es" >>> recognized by say... >>> >>> On Tue, Apr 14, 2009 at 1:57 PM, Brian West >>> wrote: >>> > This also requires you to write all the phrase macros for voicemail, >>> ivr and >>> > other things in the demo in lang/en/ >>> > /b >>> > On Apr 14, 2009, at 12:48 PM, Jo?o Mesquita wrote: >>> > >>> > I know spanish and I would translate it no problem. MC, get in touch >>> with me >>> > off-list so we can handle that. >>> > I can also translate to portuguese-brazil. >>> > jmesquita >>> > >>> > Brian West >>> > brian at freeswitch.org >>> > -- Meet us at ClueCon! http://www.cluecon.com >>> > >>> > >>> > >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at 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://blog.krisk.org >>> http://www.submityoursip.com >>> http://www.astlinux.org >>> http://www.star2star.com >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/12169928/attachment-0001.html From kristian.kielhofner at gmail.com Tue Apr 14 12:58:54 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 14 Apr 2009 15:58:54 -0400 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> Message-ID: <2d9149cd0904141258i24988b8iadb4cf87e8ce073f@mail.gmail.com> On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins wrote: > Cool. We've had several volunteers start translating the phrase files into > Spanish and Brazilian Portugese. We'll keep you posted when we have the > Spanish one ready. FYI, I committed a stub phrase_es.xml file but it hasn't > been translated yet except for the first twenty digits. However, there > aren't any audio files associated with it yet... > > -MC > Is there anything I can do with this file now? I can't seem to find the relationship between any of the phrase files in that directory and my running configuration. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From mfedyk at mikefedyk.com Tue Apr 14 20:59:54 2009 From: mfedyk at mikefedyk.com (Mike Fedyk) Date: Tue, 14 Apr 2009 20:59:54 -0700 Subject: [Freeswitch-users] Recommended tools for creating/extending a sip test suite? Message-ID: <93cdabd20904142059h37091d46lc40e571f21553f91@mail.gmail.com> Hi all, I'm looking for suggestions on which open source tools to use for creating (or extending if there is already a project for this) a sip test suite. I have already heard of sipp, but I want to know what others are using and how they go about this before starting from scratch myself. Some things I'd like to do: - Dialplan/ voice menu/provider/did testing: Call number, press 1, expect to receive call on another extension. (kinda like expect) - Load testing Basically I want to be able to automate how a human may interact with my installation to reproduce bugs and make sure they don't come back. That way I can make sure my changes (wherever they may be in my stack, dialplan, freeswitch, openser/kamailio/opensips, etc.). Any pointers and/or tips will be much appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/5ee6118b/attachment.html From mitul at enterux.com Wed Apr 15 00:22:50 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 15 Apr 2009 03:22:50 -0400 Subject: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ? Message-ID: <27149.1239780170@enterux.com> BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; }Hello there, In my previous encounter with FreeSwitch, I had found that Bret had posted on the Mailing List somewhere about availability of the entire FreeSwitch Wiki Documentation on a single PDF, this is useful coz at the offset apart from Wiki there is no other offline media to learn it. Is the same PDF available looking at the growth of Wiki pages and the updation. I look forward to hear from you guys, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions, The Enterprise Linux Company (TM), www.enterux.com +91-9820332422 ------------------------- Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090415/20372f3f/attachment.html From stevecrozz at gmail.com Tue Apr 14 22:39:31 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Tue, 14 Apr 2009 22:39:31 -0700 Subject: [Freeswitch-users] Recommended tools for creating/extending a sip test suite? In-Reply-To: <93cdabd20904142059h37091d46lc40e571f21553f91@mail.gmail.com> References: <93cdabd20904142059h37091d46lc40e571f21553f91@mail.gmail.com> Message-ID: <11990ade0904142239j757fca09jac61837c313c714c@mail.gmail.com> It seems to me like the freeswitch platform itself would be a good place to start. I haven't thoroughly thought this out, but maybe you could write a test library using mod_ designed to do human-like things such as issuing dtmf tones, pausing, speaking, etc. You could even run test scripts using the event socket (api commands) and test the results by subscribing to related events. I'd love to hear about what you come up with. --Stephen On Tue, Apr 14, 2009 at 8:59 PM, Mike Fedyk wrote: > Hi all, > > I'm looking for suggestions on which open source tools to use for creating > (or extending if there is already a project for this) a sip test suite. > > I have already heard of sipp, but I want to know what others are using and > how they go about this before starting from scratch myself. > > Some things I'd like to do: > - Dialplan/ voice menu/provider/did testing: Call number, press 1, expect > to receive call on another extension. (kinda like expect) > - Load testing > > Basically I want to be able to automate how a human may interact with my > installation to reproduce bugs and make sure they don't come back. That way > I can make sure my changes (wherever they may be in my stack, dialplan, > freeswitch, openser/kamailio/opensips, etc.). > > Any pointers and/or tips will be much appreciated. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/5c14e419/attachment.html From yudha2008 at gmail.com Wed Apr 15 00:15:52 2009 From: yudha2008 at gmail.com (Baskar) Date: Wed, 15 Apr 2009 12:45:52 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: <49E4883D.1030305@gcdf.pl> References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> <49E47D67.2080608@gcdf.pl> <49E4883D.1030305@gcdf.pl> Message-ID: *Hi, Now i can able to load the mod_java in the freeswitch console. After that i have followed these method to run the PhoneTest.java * *1) verified my classpath in the java.conf.xml: ] However, none of the files in conf have a tag called . All files are conforming xml. I can't seem to find what's changed. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/b8b33f1d/attachment.html From brian at freeswitch.org Wed Apr 29 08:32:55 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 29 Apr 2009 10:32:55 -0500 Subject: [Freeswitch-users] Very confusing startup error In-Reply-To: <98a86adf0904290821s7f7314e9laef0029115016441@mail.gmail.com> References: <98a86adf0904290821s7f7314e9laef0029115016441@mail.gmail.com> Message-ID: <9B4982CF-1E5B-4297-87EE-F76CD2AE4321@freeswitch.org> The first is an error that is unrelated to the second error. Check out freeswitch.xml.fsxml line 2423 you'll have an extra line there. /b On Apr 29, 2009, at 10:21 AM, Gerry Hull wrote: > All of a sudden I'm getting this startup error when I start > FreeSwitch: > > C:\DVLP\FreeSwitch>freeswitch > Error including C:\DVLP\FreeSwitch\conf\autoload_configs\.. > \sip_profiles\internal/*.xml (Invalid argument) > Cannot Initialize [[error near line 2423]: unexpected closing tag context>] > > However, none of the files in conf have a tag called . > All files are conforming xml. I can't seem to find what's changed. > > Any ideas? > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/fd782cc3/attachment.html From gk at exram.de Wed Apr 29 08:36:13 2009 From: gk at exram.de (Guido Kuth) Date: Wed, 29 Apr 2009 15:36:13 +0000 Subject: [Freeswitch-users] Very confusing startup error Message-ID: At least your dialplan should have a tag named . See default dialplan ! Original Message processed by David.InfoCenter Subject: [Freeswitch-users] Very confusing startup error (29-Apr-2009 17:26) From: Gerry Hull To: gk at exram.de All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitch>freeswitch Error including C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml (Invalid argument) Cannot Initialize [[error near line 2423]: unexpected closing tag ] However, none of the files in conf have a tag called . All files are conforming xml. I can't seem to find what's changed. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/ae5db11f/attachment-0001.html From gerry at pstn2.net Wed Apr 29 08:54:14 2009 From: gerry at pstn2.net (Gerry Hull) Date: Wed, 29 Apr 2009 11:54:14 -0400 Subject: [Freeswitch-users] Very confusing startup error In-Reply-To: References: Message-ID: <98a86adf0904290854v44ca7491qcff74454b3599b46@mail.gmail.com> Thanks Guys! I could not find my problem -- but you pointed me in the correct direction. I had a mismatched tag in my public.xml in the dialpan. So, is freeswitch.xml.fsxml a logged representation of the complete config file in memory? On Wed, Apr 29, 2009 at 11:36 AM, Guido Kuth wrote: > At least your dialplan should have a tag named . See default > dialplan ! > > > > Original Message > * processed by David.InfoCenter* > Subject: > [Freeswitch-users] Very confusing startup error (29-Apr-2009 17:26) > From: > Gerry Hull > To: > gk at exram.de > > All of a sudden I'm getting this startup error when I start FreeSwitch: > > C:\DVLP\FreeSwitch>freeswitch > Error including > C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml > (Invalid argument) > Cannot Initialize [[error near line 2423]: unexpected closing tag > ] > > However, none of the files in conf have a tag called . All > files are conforming xml. I can't seem to find what's changed. > > Any ideas? > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/68be6acc/attachment.html From q.edward at gmail.com Wed Apr 29 09:12:16 2009 From: q.edward at gmail.com (Edward Q.) Date: Wed, 29 Apr 2009 12:12:16 -0400 Subject: [Freeswitch-users] HELP 3-way network access In-Reply-To: <89313a90904290902p2146f19co86695c3576073b5d@mail.gmail.com> References: <89313a90904290902p2146f19co86695c3576073b5d@mail.gmail.com> Message-ID: <89313a90904290912n171c867evd1368c26cfc01ade@mail.gmail.com> Hi guys .. I need your help please... I am trying to setup an FS box. It has to be like a 3 way thing since i reside in one network - my FS machine resides on another network - and my provider (gateway) resides on another network. I am going to try to be specific as much as i can. I did a Quick and Dirty install. Here is my testing servers info. Hardware Intel Dual Core 2.6 GHZ Real memory 3.56 GB total, 267.71 MB used Hard Drives 1 SATA 250GB MotherBoard Biostar P4M900-M4 Motherboard - VIA P4M900, Socket 478, MicroATX Software Operating system CentOS Linux 5.2 Kernel and CPU Linux 2.6.18-92.1.22.el5 on i686 Apache 2.2.3 MySQL 5.0.45 SSH OpenSSH 4.3 freeswitch at internal> version FreeSWITCH Version 1.0.trunk (13181M) Ok the FS testing server resides on xxx.9.10.xxx. The gateway resides on xxx.9.9.xxx. And my computer resides on 75.74.xxx.xxx (My computer has X-lite) installed. When i create the SIP profile on X-Lite in my computer and tell X-Lite to register on xxx.9.10.xxx It says discovering network ... Initializing... Registering ... And then it shows up your Your username is: 1000 (looks like it is registered). Now when i try to dial 5000 to listen at least to the IVR demo i get ... The person you are calling is unavailable please try again ... message. and shows on the top of the username ... Call failed: Request Timeout (message) And on the fs_cli console it shows this .. freeswitch at internal> 2009-04-29 11:46:05 [DEBUG] sofia.c:4242 sofia_handle_sip_i_invite() IP 75.74.xxx.xxx Rejected by acl "domains". Falling back to Digest auth. I am a total noob on this. I replaced my original acl.conf.xml with this... I shutdown FS and then restart FS with the -nc option. And Still the same thing. My gateway is a CANTATA switch which does not require authentication. I am trying to generate a call from my 75.74.xxx.xxx using X-Lite to a PSTN phone on the outside using the CANTATA switch on xxx.9.9.xxx through my FS box on xxx.9.10.xxx But as for now I can't even get the IVR to work for now ... Since i don't know anything about FS i would like to know what am i doing wrong.. And what files have to be either created or updated to do this. Thanks to everyone for all the help Edward -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/46a04212/attachment.html From gallo at mctelefonia.com Wed Apr 29 09:18:04 2009 From: gallo at mctelefonia.com (Antonio Gallo) Date: Wed, 29 Apr 2009 18:18:04 +0200 Subject: [Freeswitch-users] is there something like "flash operator panel" for freeswitch? In-Reply-To: <191c3a030904290537p6f460642ucedd85b3856374e2@mail.gmail.com> References: <49F5B799.80809@mctelefonia.com> <191c3a030904270720j78438eecr6ae3418358d7e727@mail.gmail.com> <49F5C4A9.7000506@mctelefonia.com> <2D0785C2-9CC1-46E7-9EEE-55E49D63CB2C@freeswitch.org> <49F5C7D5.2090003@mctelefonia.com> <49F7FB59.1090608@mctelefonia.com> <191c3a030904290537p6f460642ucedd85b3856374e2@mail.gmail.com> Message-ID: <49F87DBC.7010203@mctelefonia.com> ok i did some test today using the yesterday's trunk with a gxp2010 and a snom360 both with 2 LEDS monitoring each other and themselves. Configuration: gxp2010 user: 1000 led1: 1000 led2: 1001 snom360 user: 1001 led1: 1000 led2: 1001 Problem with both phones: - when a phone reboot and it subscribe it does not get notified of the current status of the subscribed phones i.e. if gxp is on the phone the snom led1 is off/unlit i.e. if snom is on the phone the gxp led2 is off/unlit Problem with GXP only: - both subscribe LED stop working after a 1 or 2 calls until the gxp re-subscribe or re-register To skip using LED on phones is there is something like "flash operator panel" to display telephone status? Thanks in advance, Antonio (AGX) From brian at freeswitch.org Wed Apr 29 09:19:02 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 29 Apr 2009 11:19:02 -0500 Subject: [Freeswitch-users] HELP 3-way network access In-Reply-To: <89313a90904290912n171c867evd1368c26cfc01ade@mail.gmail.com> References: <89313a90904290902p2146f19co86695c3576073b5d@mail.gmail.com> <89313a90904290912n171c867evd1368c26cfc01ade@mail.gmail.com> Message-ID: <665E076F-A898-47E5-920C-2BBDA75C19AB@freeswitch.org> Now you need to open up the sofia profile in sip_profile/internal.xml and apply the test1 acl instead of the "domains" acl. /b On Apr 29, 2009, at 11:12 AM, Edward Q. wrote: > freeswitch at internal> 2009-04-29 11:46:05 [DEBUG] sofia.c:4242 > sofia_handle_sip_i_invite() IP 75.74.xxx.xxx Rejected by acl > "domains". Falling back to Digest auth. > > I am a total noob on this. I replaced my original acl.conf.xml with > this... > > > > > > > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/17b5afa3/attachment.html From msc at freeswitch.org Wed Apr 29 09:37:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Apr 2009 09:37:19 -0700 Subject: [Freeswitch-users] Very confusing startup error In-Reply-To: <98a86adf0904290854v44ca7491qcff74454b3599b46@mail.gmail.com> References: <98a86adf0904290854v44ca7491qcff74454b3599b46@mail.gmail.com> Message-ID: <87f2f3b90904290937j51e51a8ew1dfc43b1e0d1675f@mail.gmail.com> On Wed, Apr 29, 2009 at 8:54 AM, Gerry Hull wrote: > Thanks Guys! > > I could not find my problem -- but you pointed me in the correct > direction. I had a mismatched tag in my public.xml in the dialpan. > > So, is freeswitch.xml.fsxml a logged representation of the complete config > file in memory? > > Affirmative. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/99679635/attachment.html From Prometheus001 at gmx.net Wed Apr 29 10:03:45 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 29 Apr 2009 19:03:45 +0200 Subject: [Freeswitch-users] First Linux soft phone running TLS/SRTP on Linux (Zoipe Bizz) Message-ID: <49F88871.8000409@gmx.net> After 6 months of discussions with Attractel, today we finally got a new version of Zoiper Bizz, which works with TLS and SRTP (previous versions only supported TLS). I have added the info, how to set it up, in the wiki http://wiki.freeswitch.org/wiki/Interop_List#Zoiper_Bizz_2.10_and_TLS.2FSRTP We've been searching for a long time to have a working secure VoIP client under Linux. So far Zoiper seems to be the only VoIP soft phone capable of managing TLS/SRTP with Freeswitch under Linux. BTW: The free version does not support encryption. The Zoiper "Bizz" version does, but is not for free. Best regards Peter From brian at freeswitch.org Wed Apr 29 10:12:31 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 29 Apr 2009 12:12:31 -0500 Subject: [Freeswitch-users] First Linux soft phone running TLS/SRTP on Linux (Zoipe Bizz) In-Reply-To: <49F88871.8000409@gmx.net> References: <49F88871.8000409@gmx.net> Message-ID: <66B1E03E-0C70-4E91-8DA2-5067E5A6DA17@freeswitch.org> Lets not forget FreeSWITCH is a soft phone also that could do TLS and SRTP too :) /b On Apr 29, 2009, at 12:03 PM, Peter P GMX wrote: > So far Zoiper seems to be the only VoIP soft phone capable of managing > TLS/SRTP with Freeswitch under Linux. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/da43426f/attachment.html From gk at exram.de Wed Apr 29 10:21:54 2009 From: gk at exram.de (Guido Kuth) Date: Wed, 29 Apr 2009 17:21:54 +0000 Subject: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL Message-ID: I have a problem I am trying to solve for several days now. I have FS 1.3.0 installed. I have the default configuration except that I have edited event_socket.conf to match my configuration. I have two computers with x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from 1000 and my esl app answers the call plays back a greeting and after that sends a record_session command and a start_dtmf command. Now I send the bridge command with sofia/internal/1001 at ip-address. The x-lite 1001 rings and I can take the call the two can talk to each other and both are able to end the call by hanging up the phone, but there is no reaction on any dtmf tone except when I press * and 1-3, cause this is defined by bind-meta-app in default dialplan. What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Any help or suggestion is welcome. Thanks in advance...Guido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/545d60e1/attachment.html From brian at freeswitch.org Wed Apr 29 10:30:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 29 Apr 2009 12:30:21 -0500 Subject: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL In-Reply-To: References: Message-ID: If you subscribe to the event you will receive one on every DTMF press if FreeSWITCH gets it... if you happen to be getting them via inband you won't receive an event unless you enable the inband detection app. http://wiki.freeswitch.org/wiki/Event_list#DTMF http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf I also highly recommend you update to SVN trunk. /b On Apr 29, 2009, at 12:21 PM, Guido Kuth wrote: > What I need is that I get an Event on DTMF Entry on the bridged > call. Please I have to resolve this, cause this is the reason why I > came from Asterisk to FreeSwitch. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/de5dc2ca/attachment.html From anthony.minessale at gmail.com Wed Apr 29 10:39:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 29 Apr 2009 12:39:46 -0500 Subject: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL In-Reply-To: References: Message-ID: <191c3a030904291039u1501410bgbd15e422b9c3b916@mail.gmail.com> set the async flag on the socket app call that triggers your ESL connection On Wed, Apr 29, 2009 at 12:21 PM, Guido Kuth wrote: > I have a problem I am trying to solve for several days now. I have FS > 1.3.0 installed. I have the default configuration except that I have edited > event_socket.conf to match my configuration. I have two computers with > x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from > 1000 and my esl app answers the call plays back a greeting and after that > sends a record_session command and a start_dtmf command. > > Now I send the bridge command with sofia/internal/1001 at ip-address. The > x-lite 1001 rings and I can take the call the two can talk to each other and > both are able to end the call by hanging up the phone, but there is no > reaction on any dtmf tone except when I press * and 1-3, cause this is > defined by bind-meta-app in default dialplan. > > What I need is that I get an Event on DTMF Entry on the bridged call. > Please I have to resolve this, cause this is the reason why I came from > Asterisk to FreeSwitch. > > Any help or suggestion is welcome. > > Thanks in advance...Guido > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/c9daf86d/attachment.html From gk at exram.de Wed Apr 29 10:48:32 2009 From: gk at exram.de (Guido Kuth) Date: Wed, 29 Apr 2009 17:48:32 +0000 Subject: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL Message-ID: First thanks for your reply. I have subscribed to all Events, so this can't be the mistake. I sent start_dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to "info"? I think setting it to "rfc2833" would not be very meanigfull. I will try to update to svn trunk tomorrow. Again thanks for first help...Guido Original Message processed by David.InfoCenter Subject: Re: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:34) From: Brian West To: gk at exram.de If you subscribe to the event you will receive one on every DTMF press if FreeSWITCH gets it... if you happen to be getting them via inband you won't receive an event unless you enable the inband detection app. http://wiki.freeswitch.org/wiki/Event_list#DTMF http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf I also highly recommend you update to SVN trunk. /b On Apr 29, 2009, at 12:21 PM, Guido Kuth wrote: What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/add9514a/attachment-0001.html From gk at exram.de Wed Apr 29 10:52:21 2009 From: gk at exram.de (Guido Kuth) Date: Wed, 29 Apr 2009 17:52:21 +0000 Subject: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL Message-ID: Hello Anthony, sorry, but I forgot to tell you that I have an inbound ESL connection not an outbound one. So I connect to FS and then wait for Events. I know that I can set async flag in outbound socket, but is this also possible for inbound socket, and when, is it the same as in outbound socket behind the IP-Address? Thank you very much...Guido Original Message processed by David.InfoCenter Subject: Re: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:47) From: Anthony Minessale To: gk at exram.de set the async flag on the socket app call that triggers your ESL connection On Wed, Apr 29, 2009 at 12:21 PM, Guido Kuth wrote: I have a problem I am trying to solve for several days now. I have FS 1.3.0 installed. I have the default configuration except that I have edited event_socket.conf to match my configuration. I have two computers with x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from 1000 and my esl app answers the call plays back a greeting and after that sends a record_session command and a start_dtmf command. Now I send the bridge command with sofia/internal/1001 at ip-address. The x-lite 1001 rings and I can take the call the two can talk to each other and both are able to end the call by hanging up the phone, but there is no reaction on any dtmf tone except when I press * and 1-3, cause this is defined by bind-meta-app in default dialplan. What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Any help or suggestion is welcome. Thanks in advance...Guido _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/7a6e0dab/attachment.html From brian at freeswitch.org Wed Apr 29 10:54:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 29 Apr 2009 12:54:39 -0500 Subject: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL In-Reply-To: References: Message-ID: Well the best option is to NOT use inband at all if possible. And use RFC2833 which eyebeam/xlite support as do most providers out there... You do not HAVE to start_dtmf on sip channels unless they only send the DTMF inband. set the dtmf-type back to rfc2833 and restart FS. /b On Apr 29, 2009, at 12:48 PM, Guido Kuth wrote: > First thanks for your reply. > > I have subscribed to all Events, so this can't be the mistake. I > sent start_dtmf app to FreeSwitch in caller channel and the wiki > says that you have to do this on sip channels to enable inband dtmf. > I checked sofia.conf and I have found that param dtmf-type is > commented out. Would it be helpful to set this to "info"? I think > setting it to "rfc2833" would not be very meanigfull. > > I will try to update to svn trunk tomorrow. > > Again thanks for first help...Guido Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/39d7dfed/attachment.html From paul.degt at gmail.com Wed Apr 29 12:15:28 2009 From: paul.degt at gmail.com (paul.degt) Date: Wed, 29 Apr 2009 15:15:28 -0400 Subject: [Freeswitch-users] Phones become unreachable after some time Message-ID: <49F8A750.7030906@gmail.com> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds to Mysql DB for SIP registrations, presence etc. I noticed that after some time probably >30 min. phones which have been registered but without making calls become unreachable. Meaning that any call to such extension gets forwarded to VM as if it was offline, until I reload such phone. I did try to make the phones to register every 5 min. but it does not help. I also see valid registration information in sip_registrations table. X-Lite has r-port and keep alive settings on. Would appreciate any hints on what can be the issue here. From gk at exram.de Wed Apr 29 12:20:42 2009 From: gk at exram.de (Guido Kuth) Date: Wed, 29 Apr 2009 19:20:42 +0000 Subject: [Freeswitch-users] Re-2: Re: Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL Message-ID: <0002D0FE.49F8C4AA@192.168.49.2> Thank you again Brian. The reason why I want to test with inband dtmf is that in the real environment FS will be behind a conventional ISDN PBX which will work as a gateway to the ISDN Network. So I do not know if the PBX will do something like a translation between DTMF Tones to rfc and backwars. If you have experience with this any help will be very welcome....Guido Original Message processed by David InfoCenter Subject: Re: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:59) From: Brian West To: freeswitch-users at lists.freeswitch.org Well the best option is to NOT use inband at all if possible. And use RFC2833 which eyebeam/xlite support as do most providers out there... You do not HAVE to start_dtmf on sip channels unless they only send the DTMF inband. set the dtmf-type back to rfc2833 and restart FS. /b On Apr 29, 2009, at 12:48 PM, Guido Kuth wrote: First thanks for your reply. I have subscribed to all Events, so this can't be the mistake. I sent start_ dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to "info"? I think setting it to "rfc2833" would not be very meanigfull. I will try to update to svn trunk tomorrow. Again thanks for first help...Guido Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/67e16cae/attachment-0001.html From nik.middleton at noblesolutions.co.uk Wed Apr 29 12:41:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 29 Apr 2009 20:41:53 +0100 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: <49F8A750.7030906@gmail.com> References: <49F8A750.7030906@gmail.com> Message-ID: Do the phones and FS have a firewall between them? If so, sounds like the pin hole in the fw is being closed. Alot only stay open for 4 mins Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of paul.degt Sent: 29 April 2009 20:15 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Phones become unreachable after some time I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds to Mysql DB for SIP registrations, presence etc. I noticed that after some time probably >30 min. phones which have been registered but without making calls become unreachable. Meaning that any call to such extension gets forwarded to VM as if it was offline, until I reload such phone. I did try to make the phones to register every 5 min. but it does not help. I also see valid registration information in sip_registrations table. X-Lite has r-port and keep alive settings on. Would appreciate any hints on what can be the issue here. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 paul.degt at gmail.com Wed Apr 29 12:50:12 2009 From: paul.degt at gmail.com (paul.degt) Date: Wed, 29 Apr 2009 15:50:12 -0400 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: References: <49F8A750.7030906@gmail.com> Message-ID: <49F8AF74.5040004@gmail.com> They do, but all necessary ports for FS are open. If that is fw issue, are there ways to fight with it? Nik Middleton wrote: > Do the phones and FS have a firewall between them? If so, sounds like > the pin hole in the fw is being closed. Alot only stay open for 4 mins > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 29 April 2009 20:15 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Phones become unreachable after some time > > I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds > to Mysql DB for SIP registrations, presence etc. > I noticed that after some time probably >30 min. phones which have been > registered but without making calls become unreachable. Meaning that any > > call to such extension gets forwarded to VM as if it was offline, until > I reload such phone. > I did try to make the phones to register every 5 min. but it does not > help. I also see valid registration information in sip_registrations > table. X-Lite has r-port and keep alive settings on. > Would appreciate any hints on what can be the issue here. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 nik.middleton at noblesolutions.co.uk Wed Apr 29 14:07:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 29 Apr 2009 22:07:33 +0100 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: <49F8AF74.5040004@gmail.com> References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> Message-ID: Don't know where the setting is in FS, but force them to register every 120 seconds and see if that helps Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of paul.degt Sent: 29 April 2009 20:50 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Phones become unreachable after some time They do, but all necessary ports for FS are open. If that is fw issue, are there ways to fight with it? Nik Middleton wrote: > Do the phones and FS have a firewall between them? If so, sounds like > the pin hole in the fw is being closed. Alot only stay open for 4 mins > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 29 April 2009 20:15 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Phones become unreachable after some time > > I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds > to Mysql DB for SIP registrations, presence etc. > I noticed that after some time probably >30 min. phones which have been > registered but without making calls become unreachable. Meaning that any > > call to such extension gets forwarded to VM as if it was offline, until > I reload such phone. > I did try to make the phones to register every 5 min. but it does not > help. I also see valid registration information in sip_registrations > table. X-Lite has r-port and keep alive settings on. > Would appreciate any hints on what can be the issue here. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 mikael at bjerkeland.com Wed Apr 29 14:13:05 2009 From: mikael at bjerkeland.com (Mikael Bjerkeland) Date: Wed, 29 Apr 2009 23:13:05 +0200 Subject: [Freeswitch-users] is there something like "flash operator panel" for freeswitch? In-Reply-To: <49F87DBC.7010203@mctelefonia.com> References: <49F5B799.80809@mctelefonia.com> <191c3a030904270720j78438eecr6ae3418358d7e727@mail.gmail.com> <49F5C4A9.7000506@mctelefonia.com> <2D0785C2-9CC1-46E7-9EEE-55E49D63CB2C@freeswitch.org> <49F5C7D5.2090003@mctelefonia.com> <49F7FB59.1090608@mctelefonia.com> <191c3a030904290537p6f460642ucedd85b3856374e2@mail.gmail.com> <49F87DBC.7010203@mctelefonia.com> Message-ID: None that I know of, but it should be fairly simple to create FOP for FS with the event socket. 2009/4/29 Antonio Gallo > ok i did some test today using the yesterday's trunk with a gxp2010 and > a snom360 both with 2 LEDS monitoring each other and themselves. > > Configuration: > gxp2010 user: 1000 led1: 1000 led2: 1001 > snom360 user: 1001 led1: 1000 led2: 1001 > > Problem with both phones: > - when a phone reboot and it subscribe it does not get notified of the > current status of the subscribed phones > i.e. if gxp is on the phone the snom led1 is off/unlit > i.e. if snom is on the phone the gxp led2 is off/unlit > > Problem with GXP only: > - both subscribe LED stop working after a 1 or 2 calls until the gxp > re-subscribe or re-register > > > To skip using LED on phones is there is something like "flash operator > panel" to display > telephone status? > > Thanks in advance, > Antonio (AGX) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/34f6e48a/attachment.html From msc at freeswitch.org Wed Apr 29 14:20:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Apr 2009 14:20:21 -0700 Subject: [Freeswitch-users] is there something like "flash operator panel" for freeswitch? In-Reply-To: References: <49F5B799.80809@mctelefonia.com> <191c3a030904270720j78438eecr6ae3418358d7e727@mail.gmail.com> <49F5C4A9.7000506@mctelefonia.com> <2D0785C2-9CC1-46E7-9EEE-55E49D63CB2C@freeswitch.org> <49F5C7D5.2090003@mctelefonia.com> <49F7FB59.1090608@mctelefonia.com> <191c3a030904290537p6f460642ucedd85b3856374e2@mail.gmail.com> <49F87DBC.7010203@mctelefonia.com> Message-ID: <87f2f3b90904291420r21af3c71tdbeb2df79baca489@mail.gmail.com> On Wed, Apr 29, 2009 at 2:13 PM, Mikael Bjerkeland wrote: > None that I know of, but it should be fairly simple to create FOP for FS > with the event socket. > The "fairly simple" part is actually doing it. The really hard part is finding the time/energy/inclination to do it... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/a42beb80/attachment.html From anthony.minessale at gmail.com Wed Apr 29 15:22:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 29 Apr 2009 17:22:26 -0500 Subject: [Freeswitch-users] is there something like "flash operator panel" for freeswitch? In-Reply-To: <49F87DBC.7010203@mctelefonia.com> References: <49F5B799.80809@mctelefonia.com> <191c3a030904270720j78438eecr6ae3418358d7e727@mail.gmail.com> <49F5C4A9.7000506@mctelefonia.com> <2D0785C2-9CC1-46E7-9EEE-55E49D63CB2C@freeswitch.org> <49F5C7D5.2090003@mctelefonia.com> <49F7FB59.1090608@mctelefonia.com> <191c3a030904290537p6f460642ucedd85b3856374e2@mail.gmail.com> <49F87DBC.7010203@mctelefonia.com> Message-ID: <191c3a030904291522u62e66b99wdc7ce91feb76470b@mail.gmail.com> edit autoload_configs/sofia.conf.xml in add then you will see all the sql stmts etc and you can debug your issue On Wed, Apr 29, 2009 at 11:18 AM, Antonio Gallo wrote: > ok i did some test today using the yesterday's trunk with a gxp2010 and > a snom360 both with 2 LEDS monitoring each other and themselves. > > Configuration: > gxp2010 user: 1000 led1: 1000 led2: 1001 > snom360 user: 1001 led1: 1000 led2: 1001 > > Problem with both phones: > - when a phone reboot and it subscribe it does not get notified of the > current status of the subscribed phones > i.e. if gxp is on the phone the snom led1 is off/unlit > i.e. if snom is on the phone the gxp led2 is off/unlit > > Problem with GXP only: > - both subscribe LED stop working after a 1 or 2 calls until the gxp > re-subscribe or re-register > > > To skip using LED on phones is there is something like "flash operator > panel" to display > telephone status? > > Thanks in advance, > Antonio (AGX) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/474b58f4/attachment.html From cervajs at fpf.slu.cz Thu Apr 30 00:23:47 2009 From: cervajs at fpf.slu.cz (marek cervenka) Date: Thu, 30 Apr 2009 09:23:47 +0200 (CEST) Subject: [Freeswitch-users] First Linux soft phone running TLS/SRTP on Linux (Zoipe Bizz) In-Reply-To: <49F88871.8000409@gmx.net> References: <49F88871.8000409@gmx.net> Message-ID: > After 6 months of discussions with Attractel, today we finally got a new > version of Zoiper Bizz, which works with TLS and SRTP (previous versions > only supported TLS). > I have added the info, how to set it up, in the wiki > http://wiki.freeswitch.org/wiki/Interop_List#Zoiper_Bizz_2.10_and_TLS.2FSRTP > > We've been searching for a long time to have a working secure VoIP > client under Linux. > So far Zoiper seems to be the only VoIP soft phone capable of managing > TLS/SRTP with Freeswitch under Linux. QuteCom have TLS/SRTP on its roadmap http://trac.qutecom.org/roadmap --------------------------------------- Marek Cervenka ======================================= From rossmck at mac.com Thu Apr 30 02:46:44 2009 From: rossmck at mac.com (Ross McKillop) Date: Thu, 30 Apr 2009 02:46:44 -0700 (PDT) Subject: [Freeswitch-users] Ask for name in conferencing? Message-ID: <1241084804083-2746159.post@n2.nabble.com> As a former user of the app_confcall (http://www.freeswitch.org/node/100) Asterisk module produced by FreeSWITCH I'm in the process of moving a number of Asterisk-based services to FreeSWITCH and am trying to find the equivalent of the "record name before enter" feature of app_confcall. It seems strange that the FreeSWITCH conference module doesn't include this when the one built by the FreeSWITCH developers for Asterisk does... I know I could amend conference.js to do it, however there's no point in re-inventing the wheel if there's already another way. Anyone done this with FreeSWITCH, if so, how? Regards, Ross -- View this message in context: http://n2.nabble.com/Ask-for-name-in-conferencing--tp2746159p2746159.html Sent from the freeswitch-users mailing list archive at Nabble.com. From rossmck at mac.com Thu Apr 30 04:46:52 2009 From: rossmck at mac.com (Ross McKillop) Date: Thu, 30 Apr 2009 12:46:52 +0100 Subject: [Freeswitch-users] Ask for name in conferencing? Message-ID: <7D883B3A-FA3B-418D-98A6-41DAF5078E25@mac.com> As a former user of the app_confcall (http://www.freeswitch.org/node/ 100) Asterisk module produced by FreeSWITCH I'm in the process of moving a number of Asterisk-based services to FreeSWITCH and am trying to find the equivalent of the "record name before enter" feature of app_confcall. It seems strange that the FreeSWITCH conference module doesn't include this when the one built by the FreeSWITCH developers for Asterisk does... I know I could amend conference.js to do it, however there's no point in re-inventing the wheel if there's already another way. Anyone done this with FreeSWITCH, if so, how? Regards, Ross p.s. if it appears that i've posted this twice please accept my apologies - I tried through Nabble and it failed, so i've used a proper mail client this time... From daniel at rimspace.net Wed Apr 29 22:43:16 2009 From: daniel at rimspace.net (Daniel Pittman) Date: Thu, 30 Apr 2009 15:43:16 +1000 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> Message-ID: <87fxfqk87f.fsf@rimspace.net> Pawe? Pier?cionek writes: G'day Pawe?. > boot Your kernel with "divider=10 nohz=off" options :) > > Recent kernels are tickless which basically causes all freeswitch > timers/sleeps to fire at requested microsecond intervals. With nohz > kernels You get hundred times more system calls with freeswitch :( Like Jason, I am also interested to know why the tickless kernel causes the timers to generate so much more load. I can't find anything documented anywhere, really, about the issue ? and this thread is the only thing Google turns up on the topic. I am looking to move my SIP system to FreeSwitch some time soon, if I can, but I would love to know why nohz is so hostile to FreeSwitch before I do, if possible. Regards, Daniel From brian at freeswitch.org Thu Apr 30 06:18:18 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 30 Apr 2009 08:18:18 -0500 Subject: [Freeswitch-users] Ask for name in conferencing? In-Reply-To: <7D883B3A-FA3B-418D-98A6-41DAF5078E25@mac.com> References: <7D883B3A-FA3B-418D-98A6-41DAF5078E25@mac.com> Message-ID: <683E07D2-863B-4BF5-A60E-6F0BDA53DDED@freeswitch.org> Ross, I can see no reason to have the conference module do that for you when there are so many ways to do that externally with javascript, lua or any other the other languages then you can inject the sound file into the conference on demand before you drop the participant in. I like simplicity due to the fact it has less bugs /b On Apr 30, 2009, at 6:46 AM, Ross McKillop wrote: > As a former user of the app_confcall (http://www.freeswitch.org/node/ > 100) Asterisk module produced by FreeSWITCH I'm in the process of > moving a number of Asterisk-based services to FreeSWITCH and am trying > to find the equivalent of the "record name before enter" feature of > app_confcall. > > It seems strange that the FreeSWITCH conference module doesn't include > this when the one built by the FreeSWITCH developers for Asterisk > does... I know I could amend conference.js to do it, however there's > no point in re-inventing the wheel if there's already another way. > > Anyone done this with FreeSWITCH, if so, how? > > Regards, > Ross > > p.s. if it appears that i've posted this twice please accept my > apologies - I tried through Nabble and it failed, so i've used a > proper mail client this time... Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/a3af6239/attachment.html From paul.degt at gmail.com Thu Apr 30 06:34:47 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 30 Apr 2009 09:34:47 -0400 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> Message-ID: <49F9A8F7.8050703@gmail.com> Worked for Grandstream, but not for X-Lite. Nik Middleton wrote: > Don't know where the setting is in FS, but force them to register every > 120 seconds and see if that helps > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 29 April 2009 20:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Phones become unreachable after some > time > > They do, but all necessary ports for FS are open. If that is fw issue, > are there ways to fight with it? > > Nik Middleton wrote: > >> Do the phones and FS have a firewall between them? If so, sounds like >> the pin hole in the fw is being closed. Alot only stay open for 4 >> > mins > >> Regards, >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> paul.degt >> Sent: 29 April 2009 20:15 >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Phones become unreachable after some time >> >> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds >> > > >> to Mysql DB for SIP registrations, presence etc. >> I noticed that after some time probably >30 min. phones which have >> > been > >> registered but without making calls become unreachable. Meaning that >> > any > >> call to such extension gets forwarded to VM as if it was offline, >> > until > >> I reload such phone. >> I did try to make the phones to register every 5 min. but it does not >> help. I also see valid registration information in sip_registrations >> table. X-Lite has r-port and keep alive settings on. >> Would appreciate any hints on what can be the issue here. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Richard.Lamkin at mettoni.com Thu Apr 30 07:24:21 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 30 Apr 2009 15:24:21 +0100 Subject: [Freeswitch-users] RFC compliance list, FS Road map and RFC.5411 Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804F44941@nickel.mettonigroup.com> Q1 - I have looked on the wiki and was unable to find a list of RFC's that FS is intended to comply with. The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a list of SIP protocols by name but these have no RFC number against them. Have I just missed the page?, if not is there any plan to put such a page together? Is there a SIP compliance matrix ? Q2 -This finally brings me on to my question; Are there any plans to publish an FS road map ? Even a wish list of features which users of FS vote on would be helpful. I know FS is OSS and it does fall to all and not just the core team to implement/extend the product but a road map would Q3- I have recently been looking at RFC.5411 which is a basically list of SIP RFC's. A developer or system designer like me is in the future are likely to use an RFC like it as a compliance list for SIP stack selection. Ultimately, it will be the likes of marketing men who will be looking for the one stop shop for a SIP spec who use RFC5411 as the SIP part of a product spec. Are there any plans to use RFC 5411 as a goal? Best Regards Richard Lamkin Mettoni Group UK richard.lamkin at mettonigroup.com ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/d6ed3788/attachment.html From paul.degt at gmail.com Thu Apr 30 07:45:21 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 30 Apr 2009 10:45:21 -0400 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> Message-ID: <49F9B981.707@gmail.com> Correction: 2 min. registration timeout does not work for either Grandstream 386 nor for X-Lite. Will try 1 min., but I am skeptical. Grandstream has other weird issues btw, like not getting dial tone from first attempt or sometimes giving buzzing noise instead of one. My fw is fairly old Netgear unit, would newer models be better in this area, or I need SIP-aware one? Nik Middleton wrote: > Don't know where the setting is in FS, but force them to register every > 120 seconds and see if that helps > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 29 April 2009 20:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Phones become unreachable after some > time > > They do, but all necessary ports for FS are open. If that is fw issue, > are there ways to fight with it? > > Nik Middleton wrote: > >> Do the phones and FS have a firewall between them? If so, sounds like >> the pin hole in the fw is being closed. Alot only stay open for 4 >> > mins > >> Regards, >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> paul.degt >> Sent: 29 April 2009 20:15 >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Phones become unreachable after some time >> >> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds >> > > >> to Mysql DB for SIP registrations, presence etc. >> I noticed that after some time probably >30 min. phones which have >> > been > >> registered but without making calls become unreachable. Meaning that >> > any > >> call to such extension gets forwarded to VM as if it was offline, >> > until > >> I reload such phone. >> I did try to make the phones to register every 5 min. but it does not >> help. I also see valid registration information in sip_registrations >> table. X-Lite has r-port and keep alive settings on. >> Would appreciate any hints on what can be the issue here. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 30 07:46:31 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 30 Apr 2009 07:46:31 -0700 (PDT) Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <1241015506.11362.1.camel@portable-evil> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> Message-ID: <23317579.post@talk.nabble.com> Thanks guys for the links on CF-to-IDE adaptors. -- View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23317579.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gallo at mctelefonia.com Thu Apr 30 07:51:34 2009 From: gallo at mctelefonia.com (Antonio Gallo) Date: Thu, 30 Apr 2009 16:51:34 +0200 Subject: [Freeswitch-users] RFC compliance list, FS Road map and RFC.5411 In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804F44941@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804F44941@nickel.mettonigroup.com> Message-ID: <49F9BAF6.7080001@mctelefonia.com> Richard Lamkin ha scritto: > > Q1 -- I have looked on the wiki and was unable to find a list of RFC's > that FS is intended to comply with. > > The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a > list of SIP protocols by name but these have no RFC number against them. > > Have I just missed the page?, if not is there any plan to put such a > page together? > > Is there a SIP compliance matrix ? > AFAIK it uses Sofia SIP library and this is the link to the library itself implemented stuffs http://sofia-sip.sourceforge.net/refdocs/sofia_sip_conformance.html You get all the RFC numbers you want here :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/8f16f5cc/attachment.html From rossmck at mac.com Thu Apr 30 08:20:51 2009 From: rossmck at mac.com (Ross McKillop) Date: Thu, 30 Apr 2009 08:20:51 -0700 (PDT) Subject: [Freeswitch-users] Ask for name in conferencing? In-Reply-To: <683E07D2-863B-4BF5-A60E-6F0BDA53DDED@freeswitch.org> References: <7D883B3A-FA3B-418D-98A6-41DAF5078E25@mac.com> <683E07D2-863B-4BF5-A60E-6F0BDA53DDED@freeswitch.org> Message-ID: <1241104851593-2747788.post@n2.nabble.com> Brian West wrote: > > Ross, > I can see no reason to have the conference module do that for you > when there are so many ways to do that externally with javascript, lua > or any other the other languages then you can inject the sound file > into the conference on demand before you drop the participant in. I > like simplicity due to the fact it has less bugs > > /b > Thanks for the quick response ... will do it with JS... I've just not managed to get my head around Lua yet ;) Whilst it's not an ideal language it'd be quite nice to be able to use PHP with FreeSWITCH ... I've got literally hundreds of AGIs to convert and it'd make it a lot easier ... but that's for another day and another thread. Thanks again, Ross -- View this message in context: http://n2.nabble.com/Ask-for-name-in-conferencing--tp2746986p2747788.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Thu Apr 30 08:43:27 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 30 Apr 2009 11:43:27 -0400 Subject: [Freeswitch-users] RFC compliance list, FS Road map and RFC.5411 In-Reply-To: <49F9BAF6.7080001@mctelefonia.com> References: <3181A30B8C35AB4AA8577B78DDF4613804F44941@nickel.mettonigroup.com> <49F9BAF6.7080001@mctelefonia.com> Message-ID: I can confirm that we comply with rfc 5411 in that we agree that is a list of sip specs that we may or may not honor, and that we may or may not have ever seen or read. Joking aside, the sofia list is pretty good, there are some things noted as it would be implemented in the application. If you have specific questions about things notated like that in the sofia link below, reply to this thread and we'll try to sort out if we intend or possibly do support it. Mike On Apr 30, 2009, at 10:51 AM, Antonio Gallo wrote: > Richard Lamkin ha scritto: >> >> Q1 ? I have looked on the wiki and was unable to find a list of >> RFC?s that FS is intended to comply with. >> The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a >> list of SIP protocols by name but these have no RFC number against >> them. >> Have I just missed the page?, if not is there any plan to put such >> a page together? >> Is there a SIP compliance matrix ? > AFAIK it uses Sofia SIP library and this is the link to the library > itself implemented stuffs > http://sofia-sip.sourceforge.net/refdocs/sofia_sip_conformance.html > > You get all the RFC numbers you want here :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/9b3531ce/attachment.html From Richard.Lamkin at mettoni.com Thu Apr 30 08:47:35 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 30 Apr 2009 16:47:35 +0100 Subject: [Freeswitch-users] RFC compliance list, FS Road map and RFC.5411 In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380472F995@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF461380472F995@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804F44A1C@nickel.mettonigroup.com> Sorry in my eagerness I seem to have sent my original email with a few words missing. Too much cut and paste! Q1 - I have looked on the wiki and was unable to find a list of RFC's that FS is intended to comply with. The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a list of SIP protocols by name but these have no RFC number against them. Have I just missed the page?, if not is there any plan to put such a page together? Is there a SIP compliance matrix ? Q2 -Are there any plans to publish an FS road map ? Even a wish list of features which users of FS vote on would be helpful. I know FS is OSS and it does fall to all and not just the core team to implement/extend the product but a road map would be helpful. Q3- I have recently been looking at RFC.5411 which is a basically list of SIP RFC's. A developer or system designer like me is in the future are likely to use an RFC like it as a compliance list for SIP stack selection. Ultimately, it will be the likes of marketing men who will be looking for the one stop shop for a SIP spec who use RFC5411 as the SIP part of a product spec. Are there any plans to use RFC 5411 as a goal? Best Regards Richard Lamkin Mettoni Group UK richard.lamkin at mettonigroup.com ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/29d1871a/attachment.html From anthony.minessale at gmail.com Thu Apr 30 08:55:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 30 Apr 2009 10:55:59 -0500 Subject: [Freeswitch-users] Ask for name in conferencing? In-Reply-To: <1241104851593-2747788.post@n2.nabble.com> References: <7D883B3A-FA3B-418D-98A6-41DAF5078E25@mac.com> <683E07D2-863B-4BF5-A60E-6F0BDA53DDED@freeswitch.org> <1241104851593-2747788.post@n2.nabble.com> Message-ID: <191c3a030904300855t11856208yc02328ca0af33b96@mail.gmail.com> PHP works with ESL which not entirely unlike AGI On Thu, Apr 30, 2009 at 10:20 AM, Ross McKillop wrote: > > > Brian West wrote: > > > > Ross, > > I can see no reason to have the conference module do that for you > > when there are so many ways to do that externally with javascript, lua > > or any other the other languages then you can inject the sound file > > into the conference on demand before you drop the participant in. I > > like simplicity due to the fact it has less bugs > > > > /b > > > > Thanks for the quick response ... will do it with JS... I've just not > managed to get my head around Lua yet ;) > > Whilst it's not an ideal language it'd be quite nice to be able to use PHP > with FreeSWITCH ... I've got literally hundreds of AGIs to convert and it'd > make it a lot easier ... but that's for another day and another thread. > > Thanks again, > Ross > > -- > View this message in context: > http://n2.nabble.com/Ask-for-name-in-conferencing--tp2746986p2747788.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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/4ab23277/attachment-0001.html From pawel at voiceworks.pl Thu Apr 30 11:28:23 2009 From: pawel at voiceworks.pl (=?UTF-8?Q?Pawe=C5=82_Pier=C5=9Bcionek?=) Date: Thu, 30 Apr 2009 20:28:23 +0200 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel In-Reply-To: <87fxfqk87f.fsf@rimspace.net> References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> <87fxfqk87f.fsf@rimspace.net> Message-ID: <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> Hi, With really old kernels (100Hz) if You do sleep(1ms) You sleep for 10ms on average. With enterprise kernels (250Hz) Your sleep resolution increases by a factor of 4. With fresh kernels (1000Hz) You get real 1ms timer resolution - 10fold increase compared to old kernels. With tickless You get whatever resolution You want - eg when You sleep for 100 microseconds(micro not mili) then You get exactly what You wish for. Now for reasons I do no try to understand :) there are a lot of really short sleeps and fast timers in FreeSwitch - like 100 micro(1/10th of a ms). So with CentOS such a 100 microsecond sleep cannot "fire" faster then 250 times a second. With tickless kernel same 100 microsecond sleep "fires" 10k times a second. Pawel, From brian at freeswitch.org Thu Apr 30 11:36:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 30 Apr 2009 13:36:56 -0500 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel In-Reply-To: <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> <87fxfqk87f.fsf@rimspace.net> <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> Message-ID: Then it would be recommended to not do tickless clock :P /b On Apr 30, 2009, at 1:28 PM, Pawe? Pier?cionek wrote: > Hi, > > With really old kernels (100Hz) if You do sleep(1ms) You sleep for > 10ms on average. > With enterprise kernels (250Hz) Your sleep resolution increases by a > factor of 4. > With fresh kernels (1000Hz) You get real 1ms timer resolution - > 10fold increase compared to old kernels. > > With tickless You get whatever resolution You want - eg when You > sleep for 100 microseconds(micro not mili) then You get exactly what > You wish for. > > Now for reasons I do no try to understand :) there are a lot of > really short sleeps and fast timers in FreeSwitch - like 100 > micro(1/10th of a ms). > So with CentOS such a 100 microsecond sleep cannot "fire" faster > then 250 times a second. > With tickless kernel same 100 microsecond sleep "fires" 10k times a > second. > > Pawel, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/9b0eacd4/attachment.html From mike at jerris.com Thu Apr 30 11:48:21 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 30 Apr 2009 14:48:21 -0400 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel In-Reply-To: <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> <87fxfqk87f.fsf@rimspace.net> <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> Message-ID: <636F7D02-E6E2-419F-9F96-AB2AC1A893F8@jerris.com> Can you point out any place we do sub milli second sleeps? The timer thread should be doing 1ms, I can't think of any that would be less. MIke On Apr 30, 2009, at 2:28 PM, Pawe? Pier?cionek wrote: > Hi, > > With really old kernels (100Hz) if You do sleep(1ms) You sleep for > 10ms on average. > With enterprise kernels (250Hz) Your sleep resolution increases by a > factor of 4. > With fresh kernels (1000Hz) You get real 1ms timer resolution - > 10fold increase compared to old kernels. > > With tickless You get whatever resolution You want - eg when You > sleep for 100 microseconds(micro not mili) then You get exactly what > You wish for. > > Now for reasons I do no try to understand :) there are a lot of > really short sleeps and fast timers in FreeSwitch - like 100 > micro(1/10th of a ms). > So with CentOS such a 100 microsecond sleep cannot "fire" faster > then 250 times a second. > With tickless kernel same 100 microsecond sleep "fires" 10k times a > second. > > Pawel, From nik.middleton at noblesolutions.co.uk Thu Apr 30 11:54:42 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 30 Apr 2009 19:54:42 +0100 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: <49F9A8F7.8050703@gmail.com> References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> Message-ID: Xlite may be working on the timeout FS is sending. See the following from the wiki and see if that helps, but I'm not sure In domain, set -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of paul.degt Sent: 30 April 2009 14:35 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Phones become unreachable after some time Worked for Grandstream, but not for X-Lite. Nik Middleton wrote: > Don't know where the setting is in FS, but force them to register every > 120 seconds and see if that helps > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 29 April 2009 20:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Phones become unreachable after some > time > > They do, but all necessary ports for FS are open. If that is fw issue, > are there ways to fight with it? > > Nik Middleton wrote: > >> Do the phones and FS have a firewall between them? If so, sounds like >> the pin hole in the fw is being closed. Alot only stay open for 4 >> > mins > >> Regards, >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> paul.degt >> Sent: 29 April 2009 20:15 >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Phones become unreachable after some time >> >> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds >> > > >> to Mysql DB for SIP registrations, presence etc. >> I noticed that after some time probably >30 min. phones which have >> > been > >> registered but without making calls become unreachable. Meaning that >> > any > >> call to such extension gets forwarded to VM as if it was offline, >> > until > >> I reload such phone. >> I did try to make the phones to register every 5 min. but it does not >> help. I also see valid registration information in sip_registrations >> table. X-Lite has r-port and keep alive settings on. >> Would appreciate any hints on what can be the issue here. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 gallo at mctelefonia.com Thu Apr 30 12:47:57 2009 From: gallo at mctelefonia.com (Antonio Gallo - MC) Date: Thu, 30 Apr 2009 21:47:57 +0200 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> Message-ID: <49FA006D.8080200@mctelefonia.com> Nik Middleton ha scritto: > Xlite may be working on the timeout FS is sending. > If Xlite is monitoring any user when this user reboot/unregister (then Xlite get a publish_out event) when that user come back online then Xlite is never notified until it re-register/or re-start Anyway i don't care much about Xlite i just used it for testing from home with the office machine. From chris at fowler.cc Thu Apr 30 13:54:01 2009 From: chris at fowler.cc (Chris Fowler) Date: Thu, 30 Apr 2009 16:54:01 -0400 Subject: [Freeswitch-users] Audio delay when conferencing In-Reply-To: <49FA006D.8080200@mctelefonia.com> References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> <49FA006D.8080200@mctelefonia.com> Message-ID: <7454A296C7EDE34EA57199FAA401E2F114DB7AE069@VMBX113.ihostexchange.net> I'm using FreeSWITCH (Build 13168M) and we're having intermittent multi-second delays on conference bridges with more than three participants (this is not a new issue - just bubbled to the top of the stack to address). The server is running on Amazon's AWS c1.medium instance, CentOS 5.0 with Kernel 2.6.18 32-bit i386. I recorded a conference which shows the problem nicely: http://cfowl.postinbox.com/c.wav Callers are coming on via the internal sofia profile from various physical locations. I'm not sure how to proceed with debugging this issue - advice welcome. Thanks, Chris. From paul.degt at gmail.com Thu Apr 30 14:28:35 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 30 Apr 2009 17:28:35 -0400 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> Message-ID: <49FA1803.8050601@gmail.com> Somehow 60 sec. interval works on both phones just fine. Appreciate everybody's input. Nik Middleton wrote: > Xlite may be working on the timeout FS is sending. > > See the following from the wiki and see if that helps, but I'm not sure > > > In domain, set > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 30 April 2009 14:35 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Phones become unreachable after some > time > > Worked for Grandstream, but not for X-Lite. > > Nik Middleton wrote: > >> Don't know where the setting is in FS, but force them to register >> > every > >> 120 seconds and see if that helps >> >> Regards, >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> paul.degt >> Sent: 29 April 2009 20:50 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Phones become unreachable after some >> time >> >> They do, but all necessary ports for FS are open. If that is fw issue, >> > > >> are there ways to fight with it? >> >> Nik Middleton wrote: >> >> >>> Do the phones and FS have a firewall between them? If so, sounds >>> > like > >>> the pin hole in the fw is being closed. Alot only stay open for 4 >>> >>> >> mins >> >> >>> Regards, >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>> paul.degt >>> Sent: 29 April 2009 20:15 >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: [Freeswitch-users] Phones become unreachable after some time >>> >>> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS >>> > binds > >>> >>> >> >> >>> to Mysql DB for SIP registrations, presence etc. >>> I noticed that after some time probably >30 min. phones which have >>> >>> >> been >> >> >>> registered but without making calls become unreachable. Meaning that >>> >>> >> any >> >> >>> call to such extension gets forwarded to VM as if it was offline, >>> >>> >> until >> >> >>> I reload such phone. >>> I did try to make the phones to register every 5 min. but it does not >>> > > >>> help. I also see valid registration information in sip_registrations >>> table. X-Lite has r-port and keep alive settings on. >>> Would appreciate any hints on what can be the issue here. >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >>> http://www.freeswitch.org >>> >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 30 14:30:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 30 Apr 2009 16:30:43 -0500 Subject: [Freeswitch-users] Audio delay when conferencing In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F114DB7AE069@VMBX113.ihostexchange.net> References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> <49FA006D.8080200@mctelefonia.com> <7454A296C7EDE34EA57199FAA401E2F114DB7AE069@VMBX113.ihostexchange.net> Message-ID: <191c3a030904301430q579b44fex3165e10bc5b71a9d@mail.gmail.com> the mailing list is not the correct place to report issues. http://jira.freeswitch.org We have 8 people conferences that last as long as 12 hours a day every day and there is no delay. Be advised if you open a jira it will require that you download and compile and retest your issue on SVN trunk. If we cannot find an issue in FreeSWITCH, consider contacting FreeSWITCH Solutions for commercial support debugging your 3rd party elements: http://www.freeswitchsolutions.com On Thu, Apr 30, 2009 at 3:54 PM, Chris Fowler wrote: > I'm using FreeSWITCH (Build 13168M) and we're having intermittent > multi-second delays on conference bridges with more than three participants > (this is not a new issue - just bubbled to the top of the stack to address). > > The server is running on Amazon's AWS c1.medium instance, CentOS 5.0 with > Kernel 2.6.18 32-bit i386. > > I recorded a conference which shows the problem nicely: > http://cfowl.postinbox.com/c.wav > > Callers are coming on via the internal sofia profile from various physical > locations. I'm not sure how to proceed with debugging this issue - advice > welcome. > > > > > > > > > > > > > > > > value="tone_stream://%(500,0,300,200,100,50,25)"/> > > > > value="conference/conf-is-unlocked.wav"/> > > > > > > > > > Thanks, Chris. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/f2008b39/attachment.html From brian at freeswitch.org Thu Apr 30 14:32:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 30 Apr 2009 16:32:51 -0500 Subject: [Freeswitch-users] Audio delay when conferencing In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F114DB7AE069@VMBX113.ihostexchange.net> References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> <49FA006D.8080200@mctelefonia.com> <7454A296C7EDE34EA57199FAA401E2F114DB7AE069@VMBX113.ihostexchange.net> Message-ID: <7585F4EF-E4FD-4E71-BCDC-B76455B9FD33@freeswitch.org> Have you tried on non-ec2 installs? Maybe some setting on the EC2 instance is messing with it. Also don't hijack threads please! :) /b On Apr 30, 2009, at 3:54 PM, Chris Fowler wrote: > I'm using FreeSWITCH (Build 13168M) and we're having intermittent > multi-second delays on conference bridges with more than three > participants (this is not a new issue - just bubbled to the top of > the stack to address). > > The server is running on Amazon's AWS c1.medium instance, CentOS 5.0 > with Kernel 2.6.18 32-bit i386. > > I recorded a conference which shows the problem nicely: http://cfowl.postinbox.com/c.wav > > Callers are coming on via the internal sofia profile from various > physical locations. I'm not sure how to proceed with debugging this > issue - advice welcome. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/549ae6d8/attachment.html From can_man at gmx.de Thu Apr 30 15:37:01 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Fri, 01 May 2009 00:37:01 +0200 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error Message-ID: <20090430223701.280500@gmx.net> Hello, I am trying to get skypiax working, but I am having trouble with the sound. The calls fail with CALL FAILUREREASON 7 = Sound I/O error and I am getting the following error: ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi I am running centos 5.3 and have followed the installation guide on the wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When saving the configuration on my desktop I have set the sound card to snd_dummy. On the server the startup script load snd-dumy like this /sbin/modprobe snd-dummy enable=1. Below is the output of lsmod and the debug output from FS. It would be great if someone could help me fix my problem. Thank you very much. Best wishes, Phil -bash-3.2# lsmod Module Size Used by snd_dummy 12416 0 snd_seq_oss 32832 0 snd_seq_midi_event 7744 1 snd_seq_oss snd_seq 55200 4 snd_seq_oss,snd_seq_midi_event snd_seq_device 7120 1 snd_seq_oss snd_pcm_oss 44480 0 snd_mixer_oss 16512 1 snd_pcm_oss snd_pcm 79624 2 snd_dummy,snd_pcm_oss snd_timer 22088 2 snd_seq,snd_pcm snd 55976 8 snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 7456 1 snd snd_page_alloc 8720 1 snd_pcm freeswitch at voipserverServerFreeswitch> load mod_skypiax 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 718 ][none ][-1,-1,-1] globals.debug=0 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 720 ][none ][-1,-1,-1] globals.debug=8 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 731 ][none ][-1,-1,-1] codec-master globals.debug=8 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 734 ][none ][-1,-1,-1] globals.dialplan=XML 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 740 ][none ][-1,-1,-1] globals.context=default 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 743 ][none ][-1,-1,-1] globals.codec_string=gsm,ulaw 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 750 ][none ][-1,-1,-1] globals.codec_rates_string=8000,16000 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 723 ][none ][-1,-1,-1] globals.hold_music= 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 737 ][none ][-1,-1,-1] globals.destination=5000 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] interface_id=1 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] name=skypiax1 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized XInitThreads! 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax1 ][-1, 0, 0] CONFIGURING interface_id=1 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].destination=3101 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev 13177[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING interface_id=1 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 1407 ][skypiax1 ][-1, 0, 0] X Display ':101' opened 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype instance found with id #2097454 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 ][skypiax1 ][-1, 0, 0] In skypiax_signaling_thread_func: started, p=0x2aaab93226f8 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||OK||| 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||PROTOCOL 7||| 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||CURRENTUSERHANDLE xyzUK||| 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax1 ][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: xyzUK, skype_user: xyzUK! 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||USERSTATUS ONLINE||| 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev 13177[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev 13177[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=1, waiting 60 seconds for CURRENTUSERHANDLE==xyzUK 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev 13177[(nil)|37 ][WARNINGA 1004 ][skypiax1 ][-1, 0, 0] Interface_id=1 is now STARTED, the Skype client to which we are connected gave us the correct CURRENTUSERHANDLE (xyzUK) 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] interface_id=2 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] name=skypiax2 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized XInitThreads! 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax2 ][-1, 0, 0] CONFIGURING interface_id=2 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].destination=5000 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev 13177[(nil)|37 ][WARNINGA 950 ][skypiax2 ][-1, 0, 0] STARTING interface_id=2 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 1407 ][skypiax2 ][-1, 0, 0] X Display ':102' opened 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype instance found with id #2097454 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 ][skypiax2 ][-1, 0, 0] In skypiax_signaling_thread_func: started, p=0x2aaab9325c18 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: |||OK||| 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: |||PROTOCOL 7||| 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: |||CURRENTUSERHANDLE voipserver||| 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax2 ][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: voipserver, skype_user: voipserver! 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: |||USERSTATUS ONLINE||| 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev 13177[(nil)|37 ][NOTICA 976 ][skypiax2 ][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=2 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev 13177[(nil)|37 ][NOTICA 986 ][skypiax2 ][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=2, waiting 60 seconds for CURRENTUSERHANDLE==voipserver API CALL [load(mod_skypiax)] output: +OK 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev 13177[(nil)|37 ][WARNINGA 1004 ][skypiax2 ][-1, 0, 0] Interface_id=2 is now STARTED, the Skype client to which we are connected gave us the correct CURRENTUSERHANDLE (voipserver) 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].interface_id=1 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].X11_display=:101 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].name=skypiax1 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].context=default 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].dialplan=XML 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].destination=3101 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].context=default 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].interface_id=2 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].X11_display=:102 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].name=skypiax2 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].context=default 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].dialplan=XML 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].destination=5000 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].context=default 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_skypiax] 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 switch_loadable_module_process() Adding Endpoint 'skypiax' 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'sk' 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'skypiax' freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||USER paolofun6 PHONE_MOBILE +420775216536||| freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/external/07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel sofia/external/07771236762 at sipgate.co.uk entering state [received][100] 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 15141 15141 IN IP4 217.10.66.71 s=session c=IN IP4 217.10.66.71 t=0 0 m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 sofia_glue_tech_set_codec() Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms 160 samples 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 sofia_handle_sip_i_state() (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> CS_INIT 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal sofia/external/07771236762 at sipgate.co.uk [BREAK] 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) Running State Change CS_INIT 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State INIT 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/external/07771236762 at sipgate.co.uk SOFIA INIT 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> CS_ROUTING 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal sofia/external/07771236762 at sipgate.co.uk [BREAK] 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State INIT going to sleep 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) Running State Change CS_ROUTING 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State ROUTING 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 switch_core_standard_on_routing() sofia/external/07771236762 at sipgate.co.uk Standard ROUTING 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 07771236762->00442083324655 in context public Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing [public->skype_uri] continue=false Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) [skype_uri] destination_number(00442083324655) =~ /^(00442083324655)$/ break=on-false Dialplan: sofia/external/07771236762 at sipgate.co.uk Action bridge(skypiax/skypiax1/xyzTestUK) 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/external/07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal sofia/external/07771236762 at sipgate.co.uk [BREAK] 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State ROUTING going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) Running State Change CS_EXECUTE 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State EXECUTE 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/external/07771236762 at sipgate.co.uk Standard EXECUTE EXECUTE sofia/external/07771236762 at sipgate.co.uk bridge(skypiax/skypiax1/xyzTestUK) 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 channel_outgoing_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 585 ][ ][-1, 0, 0] globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() rev 13177[(nil)|37 ][DEBUG_SKYPE 151 ][skypiax1 ][-1, 0, 0] skypiax_codec SUCCESS 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/skypiax1/xyzTestUK [0375c668-b4a2-4364-a8c6-0a718d4f00a3] 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() rev 13177[(nil)|37 ][DEBUG_SKYPE 773 ][skypiax1 ][-1, 0, 0] Calling Skype, rdest is: xyzTestUK 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 ][skypiax1 ][-1, 0, 0] SENDING: |||SET AGC OFF|||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 ][skypiax1 ][-1, 0, 0] SENDING: |||SET AEC OFF|||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 ][skypiax1 ][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 channel_outgoing_channel() (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal skypiax/skypiax1/xyzTestUK [BREAK] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change CS_INIT 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal skypiax/skypiax1/xyzTestUK [BREAK] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() rev 13177[(nil)|37 ][DEBUG_SKYPE 182 ][skypiax1 ][-1, 0, 0] skypiax/skypiax1/xyzTestUK CHANNEL INIT 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change CS_ROUTING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() rev 13177[(nil)|37 ][DEBUG_SKYPE 257 ][skypiax1 ][-1, 0, 0] skypiax/skypiax1/xyzTestUK CHANNEL ROUTING 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal skypiax/skypiax1/xyzTestUK [BREAK] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change CS_CONSUME_MEDIA 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||AGC OFF||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||AEC OFF||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||CALL 455 STATUS UNPLACED||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 0, 0] Skype MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: UNPLACED,where: NULL! ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 371 ][skypiax1 ][-1, 3,116] skype_call: 455 is now UNPLACED ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,116] READING: |||CALL 455 STATUS ROUTING||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,116] Skype MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: ROUTING,where: NULL! 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 365 ][skypiax1 ][-1, 3,117] skype_call: 455 is now ROUTING 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: |||CALL 455 FAILUREREASON 7||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, value: 7,where: NULL! 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 201 ][skypiax1 ][-1, 3,117] Skype FAILED on skype_call 455. Let's wait for the FAILED message. 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: |||CALL 455 VAA_INPUT_STATUS FALSE||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, value: FALSE,where: NULL! 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: |||CALL 455 STATUS FAILED||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: FAILED,where: NULL! 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 334 ][skypiax1 ][-1, 3,112] we tried to call Skype on skype_call 455 and Skype has now FAILED 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 ][skypiax1 ][-1, 1,112] skype call ended 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK [CS_CONSUME_MEDIA] [NORMAL_CLEARING] 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 switch_channel_perform_hangup() Send signal skypiax/skypiax1/xyzTestUK [KILL] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 293 ][skypiax1 ][-1, 1,112] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal skypiax/skypiax1/xyzTestUK [BREAK] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 switch_ivr_originate() Originate Resulted in Error Cause: 16 [NORMAL_CLEARING] 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 audio_bridge_function() Originate Failed. Cause: NORMAL_CLEARING 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 audio_bridge_function() Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 switch_channel_perform_hangup() Send signal sofia/external/07771236762 at sipgate.co.uk [KILL] 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal sofia/external/07771236762 at sipgate.co.uk [BREAK] 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State EXECUTE going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) Running State Change CS_HANGUP 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State HANGUP 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/external/07771236762 at sipgate.co.uk hanging up, cause: NORMAL_CLEARING 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 480 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/07771236762 at sipgate.co.uk Standard HANGUP, cause: NORMAL_CLEARING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State HANGUP going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State Change CS_HANGUP -> CS_REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal sofia/external/07771236762 at sipgate.co.uk [BREAK] 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) Running State Change CS_REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 switch_core_session_reporting_state() (sofia/external/07771236762 at sipgate.co.uk) State REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change CS_HANGUP 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev 13177[(nil)|37 ][DEBUG_SKYPE 228 ][skypiax1 ][-1, 1,112] hanging up skype call: 455 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 ][skypiax1 ][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev 13177[(nil)|37 ][DEBUG_SKYPE 235 ][skypiax1 ][-1, 1,112] skypiax/skypiax1/xyzTestUK CHANNEL HANGUP 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard HANGUP, cause: NORMAL_CLEARING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change CS_HANGUP -> CS_REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal skypiax/skypiax1/xyzTestUK [BREAK] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change CS_REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard REPORTING, cause: NORMAL_CLEARING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State REPORTING going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change CS_REPORTING -> CS_DESTROY 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Locked, Waiting on external entities 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Ended 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK [CS_DESTROY] 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State DESTROY 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard DESTROY 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State DESTROY going to sleep 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 1,112] READING: |||ERROR 559 CALL: Action failed||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 91 ][skypiax1 ][-1, 1,112] Skype got ERROR: |||ERROR||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 93 ][skypiax1 ][-1, 1,110] skype_call now is DOWN 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 ][skypiax1 ][-1, 1,110] skype call ended 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 687 ][skypiax1 ][-1, 1,110] no session 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/external/07771236762 at sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 switch_core_session_reporting_state() (sofia/external/07771236762 at sipgate.co.uk) State REPORTING going to sleep 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State Change CS_REPORTING -> CS_DESTROY 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 switch_core_session_thread() Session 1 (sofia/external/07771236762 at sipgate.co.uk) Locked, Waiting on external entities 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 switch_core_session_thread() Session 1 (sofia/external/07771236762 at sipgate.co.uk) Ended 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 switch_core_session_thread() Close Channel sofia/external/07771236762 at sipgate.co.uk [CS_DESTROY] 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/external/07771236762 at sipgate.co.uk) State DESTROY 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/external/07771236762 at sipgate.co.uk Standard DESTROY 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/external/07771236762 at sipgate.co.uk) State DESTROY going to sleep -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a From anthony.minessale at gmail.com Thu Apr 30 16:02:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 30 Apr 2009 18:02:03 -0500 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error In-Reply-To: <20090430223701.280500@gmx.net> References: <20090430223701.280500@gmx.net> Message-ID: <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> if you put that info in a jira ticket http://jira.freeswitch.org and route it to skypeiax , the guy who maintains that module will see it. On Thu, Apr 30, 2009 at 5:37 PM, wrote: > > Hello, > > I am trying to get skypiax working, but I am having trouble with the sound. > The calls fail with CALL FAILUREREASON 7 = Sound I/O error and > I am getting the following error: > > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > > > I am running centos 5.3 and have followed the installation guide on the > wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When > saving > the configuration on my desktop I have set the sound card to snd_dummy. On > the server the startup script load snd-dumy like this /sbin/modprobe > snd-dummy enable=1. > Below is the output of lsmod and the debug output from FS. It would be > great if someone could help me fix my problem. > > Thank you very much. > Best wishes, > Phil > > > > > -bash-3.2# lsmod > Module Size Used by > snd_dummy 12416 0 > snd_seq_oss 32832 0 > snd_seq_midi_event 7744 1 snd_seq_oss > snd_seq 55200 4 snd_seq_oss,snd_seq_midi_event > snd_seq_device 7120 1 snd_seq_oss > snd_pcm_oss 44480 0 > snd_mixer_oss 16512 1 snd_pcm_oss > snd_pcm 79624 2 snd_dummy,snd_pcm_oss > snd_timer 22088 2 snd_seq,snd_pcm > snd 55976 8 > snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer > soundcore 7456 1 snd > snd_page_alloc 8720 1 snd_pcm > > > > freeswitch at voipserverServerFreeswitch> load mod_skypiax > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 718 ][none ][-1,-1,-1] > globals.debug=0 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 720 ][none ][-1,-1,-1] > globals.debug=8 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 731 ][none ][-1,-1,-1] codec-master > globals.debug=8 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 734 ][none ][-1,-1,-1] > globals.dialplan=XML > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 740 ][none ][-1,-1,-1] > globals.context=default > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 743 ][none ][-1,-1,-1] > globals.codec_string=gsm,ulaw > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 750 ][none ][-1,-1,-1] > globals.codec_rates_string=8000,16000 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 723 ][none ][-1,-1,-1] > globals.hold_music= > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 737 ][none ][-1,-1,-1] > globals.destination=5000 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] > interface_id=1 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] name=skypiax1 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized > XInitThreads! > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax1 ][-1, 0, 0] CONFIGURING > interface_id=1 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].destination=3101 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default > 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev > 13177[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING > interface_id=1 > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 > skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 1407 > ][skypiax1 ][-1, 0, 0] X Display ':101' opened > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype > instance found with id #2097454 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 > skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 > ][skypiax1 ][-1, 0, 0] In skypiax_signaling_thread_func: started, > p=0x2aaab93226f8 > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||OK||| > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||PROTOCOL 7||| > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||CONNSTATUS ONLINE||| > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||CURRENTUSERHANDLE xyzUK||| > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax1 ][-1, 0, 0] Skype > MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: > xyzUK, skype_user: xyzUK! > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||USERSTATUS ONLINE||| > 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev > 13177[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 > seconds to find a running Skype client and connect to its SKYPE API for > interface_id=1 > 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev > 13177[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running > Skype client, connected to its SKYPE API for interface_id=1, waiting 60 > seconds for CURRENTUSERHANDLE==xyzUK > 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev > 13177[(nil)|37 ][WARNINGA 1004 ][skypiax1 ][-1, 0, 0] Interface_id=1 > is now STARTED, the Skype client to which we are connected gave us the > correct CURRENTUSERHANDLE (xyzUK) > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] > interface_id=2 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] name=skypiax2 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized > XInitThreads! > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax2 ][-1, 0, 0] CONFIGURING > interface_id=2 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax2 ][-1, 0, 0] > interface_id=2 > globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].destination=5000 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default > 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev > 13177[(nil)|37 ][WARNINGA 950 ][skypiax2 ][-1, 0, 0] STARTING > interface_id=2 > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 > skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 1407 > ][skypiax2 ][-1, 0, 0] X Display ':102' opened > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype > instance found with id #2097454 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 > skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 > ][skypiax2 ][-1, 0, 0] In skypiax_signaling_thread_func: started, > p=0x2aaab9325c18 > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: > |||OK||| > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: > |||PROTOCOL 7||| > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: > |||CONNSTATUS ONLINE||| > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: > |||CURRENTUSERHANDLE voipserver||| > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax2 ][-1, 0, 0] Skype > MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: > voipserver, skype_user: voipserver! > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: > |||USERSTATUS ONLINE||| > 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev > 13177[(nil)|37 ][NOTICA 976 ][skypiax2 ][-1, 0, 0] WAITING roughly 10 > seconds to find a running Skype client and connect to its SKYPE API for > interface_id=2 > 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev > 13177[(nil)|37 ][NOTICA 986 ][skypiax2 ][-1, 0, 0] Found a running > Skype client, connected to its SKYPE API for interface_id=2, waiting 60 > seconds for CURRENTUSERHANDLE==voipserver > API CALL [load(mod_skypiax)] output: > +OK > > 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev > 13177[(nil)|37 ][WARNINGA 1004 ][skypiax2 ][-1, 0, 0] Interface_id=2 > is now STARTED, the Skype client to which we are connected gave us the > correct CURRENTUSERHANDLE (voipserver) > > > > > > > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].interface_id=1 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].X11_display=:101 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].name=skypiax1 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].context=default > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].dialplan=XML > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].destination=3101 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].context=default > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].interface_id=2 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].X11_display=:102 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].name=skypiax2 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].context=default > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].dialplan=XML > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].destination=5000 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].context=default > 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 > switch_loadable_module_load_file() Successfully Loaded [mod_skypiax] > 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 > switch_loadable_module_process() Adding Endpoint 'skypiax' > 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 > switch_loadable_module_process() Adding API Function 'sk' > 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 > switch_loadable_module_process() Adding API Function 'skypiax' > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] > skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 > ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||USER paolofun6 > PHONE_MOBILE +420775216536||| > > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 [NOTICE] > switch_channel.c:602 switch_channel_set_name() New Channel sofia/external/ > 07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] > 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel > sofia/external/07771236762 at sipgate.co.uk entering state [received][100] > 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() Remote > SDP: > v=0 > o=root 15141 15141 IN IP4 217.10.66.71 > s=session > c=IN IP4 217.10.66.71 > t=0 0 > m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=30 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:112 G726-32/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 sofia_glue_tech_set_codec() > Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms 160 > samples > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 sofia_handle_sip_i_state() > (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> CS_INIT > 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [BREAK] > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > Running State Change CS_INIT > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > INIT > 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/external/ > 07771236762 at sipgate.co.uk SOFIA INIT > 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() > (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> > CS_ROUTING > 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [BREAK] > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > INIT going to sleep > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > Running State Change CS_ROUTING > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > ROUTING > 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 > switch_core_standard_on_routing() sofia/external/07771236762 at sipgate.co.ukStandard ROUTING > 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing 07771236762->00442083324655 in context public > Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing > [public->skype_uri] continue=false > Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) > [skype_uri] destination_number(00442083324655) =~ /^(00442083324655)$/ > break=on-false > Dialplan: sofia/external/07771236762 at sipgate.co.uk Action > bridge(skypiax/skypiax1/xyzTestUK) > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 > switch_core_standard_on_routing() (sofia/external/ > 07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [BREAK] > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > ROUTING going to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > Running State Change CS_EXECUTE > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > EXECUTE > 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() > sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 > switch_core_standard_on_execute() sofia/external/07771236762 at sipgate.co.ukStandard EXECUTE > EXECUTE sofia/external/07771236762 at sipgate.co.ukbridge(skypiax/skypiax1/xyzTestUK) > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 channel_outgoing_channel() > rev 13177[(nil)|37 ][DEBUG_SKYPE 585 ][ ][-1, 0, 0] > globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() rev > 13177[(nil)|37 ][DEBUG_SKYPE 151 ][skypiax1 ][-1, 0, 0] skypiax_codec > SUCCESS > 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New Channel skypiax/skypiax1/xyzTestUK > [0375c668-b4a2-4364-a8c6-0a718d4f00a3] > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() rev > 13177[(nil)|37 ][DEBUG_SKYPE 773 ][skypiax1 ][-1, 0, 0] Calling > Skype, rdest is: xyzTestUK > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > ][skypiax1 ][-1, 0, 0] SENDING: |||SET AGC OFF|||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > ][skypiax1 ][-1, 0, 0] SENDING: |||SET AEC OFF|||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > ][skypiax1 ][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 channel_outgoing_channel() > (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal > skypiax/skypiax1/xyzTestUK [BREAK] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change > CS_INIT > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() > (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal > skypiax/skypiax1/xyzTestUK [BREAK] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() rev > 13177[(nil)|37 ][DEBUG_SKYPE 182 ][skypiax1 ][-1, 0, 0] > skypiax/skypiax1/xyzTestUK CHANNEL INIT > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT going to > sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change > CS_ROUTING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() rev > 13177[(nil)|37 ][DEBUG_SKYPE 257 ][skypiax1 ][-1, 0, 0] > skypiax/skypiax1/xyzTestUK CHANNEL ROUTING > 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 > originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change CS_ROUTING > -> CS_CONSUME_MEDIA > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal > skypiax/skypiax1/xyzTestUK [BREAK] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING going > to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change > CS_CONSUME_MEDIA > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||AGC OFF||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||AEC OFF||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||CALL 455 STATUS UNPLACED||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 0, 0] Skype > MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: UNPLACED,where: > NULL! > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 371 ][skypiax1 ][-1, 3,116] > skype_call: 455 is now UNPLACED > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,116] READING: > |||CALL 455 STATUS ROUTING||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,116] Skype > MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: ROUTING,where: > NULL! > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 365 ][skypiax1 ][-1, 3,117] > skype_call: 455 is now ROUTING > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: > |||CALL 455 FAILUREREASON 7||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype > MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, value: 7,where: > NULL! > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 201 ][skypiax1 ][-1, 3,117] Skype > FAILED on skype_call 455. Let's wait for the FAILED message. > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: > |||CALL 455 VAA_INPUT_STATUS FALSE||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype > MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, value: > FALSE,where: NULL! > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: > |||CALL 455 STATUS FAILED||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype > MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: FAILED,where: > NULL! > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 334 ][skypiax1 ][-1, 3,112] we tried > to call Skype on skype_call 455 and Skype has now FAILED > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 > skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 > ][skypiax1 ][-1, 1,112] skype call ended > 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 > skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK > [CS_CONSUME_MEDIA] [NORMAL_CLEARING] > 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 > switch_channel_perform_hangup() Send signal skypiax/skypiax1/xyzTestUK > [KILL] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 293 ][skypiax1 ][-1, 1,112] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal > skypiax/skypiax1/xyzTestUK [BREAK] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 > switch_ivr_originate() Originate Resulted in Error Cause: 16 > [NORMAL_CLEARING] > 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 audio_bridge_function() > Originate Failed. Cause: NORMAL_CLEARING > 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 audio_bridge_function() > Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 > switch_channel_perform_hangup() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [KILL] > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [BREAK] > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > EXECUTE going to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > Running State Change CS_HANGUP > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > HANGUP > 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel > sofia/external/07771236762 at sipgate.co.uk hanging up, cause: > NORMAL_CLEARING > 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to > INVITE with: 480 > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/external/07771236762 at sipgate.co.ukStandard HANGUP, cause: NORMAL_CLEARING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > HANGUP going to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > Change CS_HANGUP -> CS_REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [BREAK] > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > Running State Change CS_REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > switch_core_session_reporting_state() (sofia/external/ > 07771236762 at sipgate.co.uk) State REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA > going to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change > CS_HANGUP > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev > 13177[(nil)|37 ][DEBUG_SKYPE 228 ][skypiax1 ][-1, 1,112] hanging up > skype call: 455 > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > ][skypiax1 ][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev > 13177[(nil)|37 ][DEBUG_SKYPE 235 ][skypiax1 ][-1, 1,112] > skypiax/skypiax1/xyzTestUK CHANNEL HANGUP > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard HANGUP, > cause: NORMAL_CLEARING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP going to > sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change > CS_HANGUP -> CS_REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal > skypiax/skypiax1/xyzTestUK [BREAK] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change > CS_REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State > REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard > REPORTING, cause: NORMAL_CLEARING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State > REPORTING going to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change > CS_REPORTING -> CS_DESTROY > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 > switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Locked, > Waiting on external entities > 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 > switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Ended > 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 > switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK > [CS_DESTROY] > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State > DESTROY > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard > DESTROY > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State > DESTROY going to sleep > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 1,112] READING: > |||ERROR 559 CALL: Action failed||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 91 ][skypiax1 ][-1, 1,112] Skype > got ERROR: |||ERROR||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 93 ][skypiax1 ][-1, 1,110] > skype_call now is DOWN > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 > skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 > ][skypiax1 ][-1, 1,110] skype call ended > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 > skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 687 > ][skypiax1 ][-1, 1,110] no session > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() sofia/external/ > 07771236762 at sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 > switch_core_session_reporting_state() (sofia/external/ > 07771236762 at sipgate.co.uk) State REPORTING going to sleep > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > Change CS_REPORTING -> CS_DESTROY > 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 > switch_core_session_thread() Session 1 (sofia/external/ > 07771236762 at sipgate.co.uk) Locked, Waiting on external entities > 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 > switch_core_session_thread() Session 1 (sofia/external/ > 07771236762 at sipgate.co.uk) Ended > 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 > switch_core_session_thread() Close Channel sofia/external/ > 07771236762 at sipgate.co.uk [CS_DESTROY] > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/external/ > 07771236762 at sipgate.co.uk) State DESTROY > 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() > sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() sofia/external/07771236762 at sipgate.co.ukStandard DESTROY > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/external/ > 07771236762 at sipgate.co.uk) State DESTROY going to sleep > -- > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/76b8b881/attachment-0001.html From gcd at i.ph Thu Apr 30 19:08:01 2009 From: gcd at i.ph (Nandy Dagondon) Date: Fri, 1 May 2009 10:08:01 +0800 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <23317579.post@talk.nabble.com> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> Message-ID: <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> hi guys, i've installed FreeNas using CF-to-IDE adaptor and SanDisk 128MB CF. it's working fine. but i want to try FS on a 16GB Kingston CF. anyone tried this? if none, i can also settle down for 8GB. pls mention which brand/size works. tks, nandy On Thu, Apr 30, 2009 at 10:46 PM, Fred-145 wrote: > > Thanks guys for the links on CF-to-IDE adaptors. > -- > View this message in context: > http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23317579.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/20090501/e328c3c0/attachment.html From mitch.capper at gmail.com Thu Apr 30 20:31:31 2009 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 30 Apr 2009 23:31:31 -0400 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> Message-ID: You may want to look at the Intel Atom combo machines you can get a 1.6 ghz machine probably for around $100-150 USD in a very small form factor and very powerful. ~Mitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/27dcffc4/attachment.html From brian at freeswitch.org Thu Apr 30 20:40:26 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 30 Apr 2009 22:40:26 -0500 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> Message-ID: I have two intel atom boxes sitting on a shelf above my desk ... works like a charm! /b On Apr 30, 2009, at 10:31 PM, Mitch Capper wrote: > You may want to look at the Intel Atom combo machines you can get a > 1.6 ghz machine probably for around $100-150 USD in a very small > form factor and very powerful. > > ~Mitch Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/d7cfae74/attachment.html From mszlazak at aol.com Thu Apr 30 21:07:48 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Fri, 01 May 2009 00:07:48 -0400 Subject: [Freeswitch-users] Latest SVN update gives Windows Express compiler errors ... Message-ID: <8CB98298639AEA6-280-33C3@webmail-dx08.sysops.aol.com> I'm getting Windows Express compiler errors on the latest svn update to trunk 13213. It looks like the path is wrong to some files. Instead of folder "Debug", it's looking for files in folder "Debug DLL" Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/0074253d/attachment.html From gcd at i.ph Thu Apr 30 21:16:09 2009 From: gcd at i.ph (Nandy Dagondon) Date: Fri, 1 May 2009 12:16:09 +0800 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> Message-ID: <7d0bfd8c0904302116x71e1746es4c23dee52a894eed@mail.gmail.com> rhino used the dual-core atom mobo d945gclf2 but it requires downloading/building the linux r8168 LAN driver. -nandy On Fri, May 1, 2009 at 11:40 AM, Brian West wrote: > I have two intel atom boxes sitting on a shelf above my desk ... works like > a charm! > /b > > On Apr 30, 2009, at 10:31 PM, Mitch Capper wrote: > > You may want to look at the Intel Atom combo machines you can get a 1.6 > ghz machine probably for around $100-150 USD in a very small form factor and > very powerful. > > ~Mitch > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/7cf3c47e/attachment.html From brian at freeswitch.org Thu Apr 30 21:18:49 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 30 Apr 2009 23:18:49 -0500 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <7d0bfd8c0904302116x71e1746es4c23dee52a894eed@mail.gmail.com> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> <7d0bfd8c0904302116x71e1746es4c23dee52a894eed@mail.gmail.com> Message-ID: Sounds like the MSI Wind :P I had to do the same thing! /b On Apr 30, 2009, at 11:16 PM, Nandy Dagondon wrote: > rhino used the dual-core atom mobo d945gclf2 but it requires > downloading/building the linux r8168 LAN driver. > > -nandy Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/ef0fabe2/attachment-0001.html From technical at ttnc.co.uk Thu Apr 30 22:50:16 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Fri, 01 May 2009 06:50:16 +0100 Subject: [Freeswitch-users] uuid_displace & FIFO help In-Reply-To: <191c3a030904270519h6f85d391p72ca1500f94cfaa5@mail.gmail.com> References: <49F07EC5.5040504@barakatdesigns.net> <18FCD53D-A0ED-4A5A-80A7-A9C7E1FF3349@freeswitch.org> <49F0840C.7030305@ttnc.co.uk> <191c3a030904231418h6d11e11bp4e84f44ea2abf179@mail.gmail.com> <49F0E3A8.5030400@ttnc.co.uk> <49F16826.5050203@ttnc.co.uk> <6D57020E-7D08-4886-A2BC-6F139E6C1BD6@freeswitch.org> <49F1C880.5040300@ttnc.co.uk> <191c3a030904241859w4bc26e84u3d8640dda76a961e@mail.gmail.com> <49F56F3B.8000906@ttnc.co.uk> <191c3a030904270519h6f85d391p72ca1500f94cfaa5@mail.gmail.com> Message-ID: <49FA8D98.3040900@ttnc.co.uk> Anthony Minessale wrote: > Also is there any way to stop uuid_broadcast as I'd > need to stop it somehow if the destination picks up? > > break all "uuid_broadcast phrase::saynumber,1" doesn't set the 'current_application_response' variable in the same way as "uuid_broadcast playback::filename.wav" does (which my script looks for to know when to move on to the next application). I've attached a patch which sets this variable if it's any use to anyone (I'm not that great at C so I hope it's correct, any comments/improvements are welcome). Thanks again Adnan -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: mod_dptools.patch Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/095a4d23/attachment.pl From gmaruzz at celliax.org Thu Apr 30 23:20:10 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 1 May 2009 08:20:10 +0200 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error In-Reply-To: <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> References: <20090430223701.280500@gmx.net> <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> Message-ID: <7b197bef0904302320t6d025985vc4e912b4373577b1@mail.gmail.com> Have a happy MayDay! I cannot see the whole mail now, it's clipped for my mobile, but it seems the nth bizarry of new alsa config file, that creates an hdmi device even if you do not have one. Try to edit /usr/share/alsa/alsa.conf or any other file in /usr/share/alsa dir and delete any mention of 'hdmi'. If this do not works, please file a jira or write again. Giovanni On 5/1/09, Anthony Minessale wrote: > if you put that info in a jira ticket > > http://jira.freeswitch.org > > and route it to skypeiax , the guy who maintains that module will see it. > > > On Thu, Apr 30, 2009 at 5:37 PM, wrote: > >> >> Hello, >> >> I am trying to get skypiax working, but I am having trouble with the >> sound. >> The calls fail with CALL FAILUREREASON 7 = Sound I/O error and >> I am getting the following error: >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM >> cards.pcm.hdmi >> >> >> I am running centos 5.3 and have followed the installation guide on the >> wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When >> saving >> the configuration on my desktop I have set the sound card to snd_dummy. On >> the server the startup script load snd-dumy like this /sbin/modprobe >> snd-dummy enable=1. >> Below is the output of lsmod and the debug output from FS. It would be >> great if someone could help me fix my problem. >> >> Thank you very much. >> Best wishes, >> Phil >> >> >> >> >> -bash-3.2# lsmod >> Module Size Used by >> snd_dummy 12416 0 >> snd_seq_oss 32832 0 >> snd_seq_midi_event 7744 1 snd_seq_oss >> snd_seq 55200 4 snd_seq_oss,snd_seq_midi_event >> snd_seq_device 7120 1 snd_seq_oss >> snd_pcm_oss 44480 0 >> snd_mixer_oss 16512 1 snd_pcm_oss >> snd_pcm 79624 2 snd_dummy,snd_pcm_oss >> snd_timer 22088 2 snd_seq,snd_pcm >> snd 55976 8 >> snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer >> soundcore 7456 1 snd >> snd_page_alloc 8720 1 snd_pcm >> >> >> >> freeswitch at voipserverServerFreeswitch> load mod_skypiax >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 718 ][none ][-1,-1,-1] >> globals.debug=0 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 720 ][none ][-1,-1,-1] >> globals.debug=8 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 731 ][none ][-1,-1,-1] >> codec-master >> globals.debug=8 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 734 ][none ][-1,-1,-1] >> globals.dialplan=XML >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 740 ][none ][-1,-1,-1] >> globals.context=default >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 743 ][none ][-1,-1,-1] >> globals.codec_string=gsm,ulaw >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 750 ][none ][-1,-1,-1] >> globals.codec_rates_string=8000,16000 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 723 ][none ][-1,-1,-1] >> globals.hold_music= >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 737 ][none ][-1,-1,-1] >> globals.destination=5000 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] >> interface_id=1 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] >> name=skypiax1 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized >> XInitThreads! >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax1 ][-1, 0, 0] CONFIGURING >> interface_id=1 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].destination=3101 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev >> 13177[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING >> interface_id=1 >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE >> 1407 >> ][skypiax1 ][-1, 0, 0] X Display ':101' opened >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype >> instance found with id #2097454 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 >> ][skypiax1 ][-1, 0, 0] In skypiax_signaling_thread_func: started, >> p=0x2aaab93226f8 >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||OK||| >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||PROTOCOL 7||| >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||CONNSTATUS ONLINE||| >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||CURRENTUSERHANDLE xyzUK||| >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax1 ][-1, 0, 0] Skype >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, >> cuh: >> xyzUK, skype_user: xyzUK! >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||USERSTATUS ONLINE||| >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev >> 13177[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly >> 10 >> seconds to find a running Skype client and connect to its SKYPE API for >> interface_id=1 >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev >> 13177[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running >> Skype client, connected to its SKYPE API for interface_id=1, waiting 60 >> seconds for CURRENTUSERHANDLE==xyzUK >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev >> 13177[(nil)|37 ][WARNINGA 1004 ][skypiax1 ][-1, 0, 0] Interface_id=1 >> is now STARTED, the Skype client to which we are connected gave us the >> correct CURRENTUSERHANDLE (xyzUK) >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] >> interface_id=2 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] >> name=skypiax2 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized >> XInitThreads! >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax2 ][-1, 0, 0] CONFIGURING >> interface_id=2 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax2 ][-1, 0, 0] >> interface_id=2 >> globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].destination=5000 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev >> 13177[(nil)|37 ][WARNINGA 950 ][skypiax2 ][-1, 0, 0] STARTING >> interface_id=2 >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE >> 1407 >> ][skypiax2 ][-1, 0, 0] X Display ':102' opened >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype >> instance found with id #2097454 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 >> ][skypiax2 ][-1, 0, 0] In skypiax_signaling_thread_func: started, >> p=0x2aaab9325c18 >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] >> READING: >> |||OK||| >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] >> READING: >> |||PROTOCOL 7||| >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] >> READING: >> |||CONNSTATUS ONLINE||| >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] >> READING: >> |||CURRENTUSERHANDLE voipserver||| >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax2 ][-1, 0, 0] Skype >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, >> cuh: >> voipserver, skype_user: voipserver! >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] >> READING: >> |||USERSTATUS ONLINE||| >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev >> 13177[(nil)|37 ][NOTICA 976 ][skypiax2 ][-1, 0, 0] WAITING roughly >> 10 >> seconds to find a running Skype client and connect to its SKYPE API for >> interface_id=2 >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev >> 13177[(nil)|37 ][NOTICA 986 ][skypiax2 ][-1, 0, 0] Found a running >> Skype client, connected to its SKYPE API for interface_id=2, waiting 60 >> seconds for CURRENTUSERHANDLE==voipserver >> API CALL [load(mod_skypiax)] output: >> +OK >> >> 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev >> 13177[(nil)|37 ][WARNINGA 1004 ][skypiax2 ][-1, 0, 0] Interface_id=2 >> is now STARTED, the Skype client to which we are connected gave us the >> correct CURRENTUSERHANDLE (voipserver) >> >> >> >> >> >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].interface_id=1 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].X11_display=:101 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].name=skypiax1 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].context=default >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].dialplan=XML >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].destination=3101 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].context=default >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].interface_id=2 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].X11_display=:102 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].name=skypiax2 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].context=default >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].dialplan=XML >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].destination=5000 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].context=default >> 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 >> switch_loadable_module_load_file() Successfully Loaded [mod_skypiax] >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 >> switch_loadable_module_process() Adding Endpoint 'skypiax' >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 >> switch_loadable_module_process() Adding API Function 'sk' >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 >> switch_loadable_module_process() Adding API Function 'skypiax' >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] >> skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 >> ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||USER paolofun6 >> PHONE_MOBILE +420775216536||| >> >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 [NOTICE] >> switch_channel.c:602 switch_channel_set_name() New Channel sofia/external/ >> 07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() >> Channel >> sofia/external/07771236762 at sipgate.co.uk entering state [received][100] >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() Remote >> SDP: >> v=0 >> o=root 15141 15141 IN IP4 217.10.66.71 >> s=session >> c=IN IP4 217.10.66.71 >> t=0 0 >> m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=fmtp:97 mode=30 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:112 G726-32/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 sofia_glue_tech_set_codec() >> Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms 160 >> samples >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp() >> Set 2833 dtmf payload to 101 >> 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 sofia_handle_sip_i_state() >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> CS_INIT >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [BREAK] >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> Running State Change CS_INIT >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> INIT >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/external/ >> 07771236762 at sipgate.co.uk SOFIA INIT >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> >> CS_ROUTING >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [BREAK] >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> INIT going to sleep >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> Running State Change CS_ROUTING >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> ROUTING >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() >> sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 >> switch_core_standard_on_routing() >> sofia/external/07771236762 at sipgate.co.ukStandard ROUTING >> 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing 07771236762->00442083324655 in context public >> Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing >> [public->skype_uri] continue=false >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) >> [skype_uri] destination_number(00442083324655) =~ /^(00442083324655)$/ >> break=on-false >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Action >> bridge(skypiax/skypiax1/xyzTestUK) >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 >> switch_core_standard_on_routing() (sofia/external/ >> 07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [BREAK] >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> ROUTING going to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> Running State Change CS_EXECUTE >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> EXECUTE >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() >> sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 >> switch_core_standard_on_execute() >> sofia/external/07771236762 at sipgate.co.ukStandard EXECUTE >> EXECUTE >> sofia/external/07771236762 at sipgate.co.ukbridge(skypiax/skypiax1/xyzTestUK) >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 channel_outgoing_channel() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 585 ][ ][-1, 0, 0] >> globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 151 ][skypiax1 ][-1, 0, 0] >> skypiax_codec >> SUCCESS >> 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() >> New Channel skypiax/skypiax1/xyzTestUK >> [0375c668-b4a2-4364-a8c6-0a718d4f00a3] >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 773 ][skypiax1 ][-1, 0, 0] Calling >> Skype, rdest is: xyzTestUK >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 >> ][skypiax1 ][-1, 0, 0] SENDING: |||SET AGC OFF|||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 >> ][skypiax1 ][-1, 0, 0] SENDING: |||SET AEC OFF|||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 >> ][skypiax1 ][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 channel_outgoing_channel() >> (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal >> skypiax/skypiax1/xyzTestUK [BREAK] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> Change >> CS_INIT >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() >> (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal >> skypiax/skypiax1/xyzTestUK [BREAK] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 182 ][skypiax1 ][-1, 0, 0] >> skypiax/skypiax1/xyzTestUK CHANNEL INIT >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT going to >> sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> Change >> CS_ROUTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 257 ][skypiax1 ][-1, 0, 0] >> skypiax/skypiax1/xyzTestUK CHANNEL ROUTING >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 >> originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change >> CS_ROUTING >> -> CS_CONSUME_MEDIA >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal >> skypiax/skypiax1/xyzTestUK [BREAK] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING going >> to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> Change >> CS_CONSUME_MEDIA >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||AGC OFF||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||AEC OFF||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||CALL 455 STATUS UNPLACED||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 0, 0] Skype >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >> UNPLACED,where: >> NULL! >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 371 ][skypiax1 ][-1, 3,116] >> skype_call: 455 is now UNPLACED >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,116] >> READING: >> |||CALL 455 STATUS ROUTING||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,116] Skype >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >> ROUTING,where: >> NULL! >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 365 ][skypiax1 ][-1, 3,117] >> skype_call: 455 is now ROUTING >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] >> READING: >> |||CALL 455 FAILUREREASON 7||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype >> MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, value: >> 7,where: >> NULL! >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 201 ][skypiax1 ][-1, 3,117] Skype >> FAILED on skype_call 455. Let's wait for the FAILED message. >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] >> READING: >> |||CALL 455 VAA_INPUT_STATUS FALSE||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype >> MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, value: >> FALSE,where: NULL! >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] >> READING: >> |||CALL 455 STATUS FAILED||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: FAILED,where: >> NULL! >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 334 ][skypiax1 ][-1, 3,112] we >> tried >> to call Skype on skype_call 455 and Skype has now FAILED >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 >> ][skypiax1 ][-1, 1,112] skype call ended >> 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 >> skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK >> [CS_CONSUME_MEDIA] [NORMAL_CLEARING] >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 >> switch_channel_perform_hangup() Send signal skypiax/skypiax1/xyzTestUK >> [KILL] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 293 ][skypiax1 ][-1, 1,112] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal >> skypiax/skypiax1/xyzTestUK [BREAK] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 >> switch_ivr_originate() Originate Resulted in Error Cause: 16 >> [NORMAL_CLEARING] >> 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 audio_bridge_function() >> Originate Failed. Cause: NORMAL_CLEARING >> 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 audio_bridge_function() >> Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] >> [NORMAL_CLEARING] >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 >> switch_channel_perform_hangup() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [KILL] >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [BREAK] >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> EXECUTE going to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> Running State Change CS_HANGUP >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> HANGUP >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel >> sofia/external/07771236762 at sipgate.co.uk hanging up, cause: >> NORMAL_CLEARING >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding >> to >> INVITE with: 480 >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() >> sofia/external/07771236762 at sipgate.co.ukStandard HANGUP, cause: >> NORMAL_CLEARING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> HANGUP going to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> Change CS_HANGUP -> CS_REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [BREAK] >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> Running State Change CS_REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >> switch_core_session_reporting_state() (sofia/external/ >> 07771236762 at sipgate.co.uk) State REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA >> going to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> Change >> CS_HANGUP >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 228 ][skypiax1 ][-1, 1,112] hanging up >> skype call: 455 >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 >> ][skypiax1 ][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 235 ][skypiax1 ][-1, 1,112] >> skypiax/skypiax1/xyzTestUK CHANNEL HANGUP >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard >> HANGUP, >> cause: NORMAL_CLEARING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP going >> to >> sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change >> CS_HANGUP -> CS_REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal >> skypiax/skypiax1/xyzTestUK [BREAK] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> Change >> CS_REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State >> REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 >> switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard >> REPORTING, cause: NORMAL_CLEARING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State >> REPORTING going to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change >> CS_REPORTING -> CS_DESTROY >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) >> Locked, >> Waiting on external entities >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Ended >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 >> switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK >> [CS_DESTROY] >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State >> DESTROY >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 >> switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard >> DESTROY >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State >> DESTROY going to sleep >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 1,112] >> READING: >> |||ERROR 559 CALL: Action failed||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 91 ][skypiax1 ][-1, 1,112] Skype >> got ERROR: |||ERROR||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 93 ][skypiax1 ][-1, 1,110] >> skype_call now is DOWN >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 >> ][skypiax1 ][-1, 1,110] skype call ended >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 687 >> ][skypiax1 ][-1, 1,110] no session >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 >> switch_core_standard_on_reporting() sofia/external/ >> 07771236762 at sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 >> switch_core_session_reporting_state() (sofia/external/ >> 07771236762 at sipgate.co.uk) State REPORTING going to sleep >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> Change CS_REPORTING -> CS_DESTROY >> 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 >> switch_core_session_thread() Session 1 (sofia/external/ >> 07771236762 at sipgate.co.uk) Locked, Waiting on external entities >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 >> switch_core_session_thread() Session 1 (sofia/external/ >> 07771236762 at sipgate.co.uk) Ended >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 >> switch_core_session_thread() Close Channel sofia/external/ >> 07771236762 at sipgate.co.uk [CS_DESTROY] >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/external/ >> 07771236762 at sipgate.co.uk) State DESTROY >> 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >> sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 >> switch_core_standard_on_destroy() >> sofia/external/07771236762 at sipgate.co.ukStandard DESTROY >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/external/ >> 07771236762 at sipgate.co.uk) State DESTROY going to sleep >> -- >> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + >> Telefonanschluss f?r nur 17,95 Euro/mtl.!* >> http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 From ribs at acm.org Wed Apr 1 00:00:53 2009 From: ribs at acm.org (Larry Edelstein) Date: Wed, 1 Apr 2009 00:00:53 -0700 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> Message-ID: <2021f8b20904010000i11611a1ewa4577d6d97ce9ba8@mail.gmail.com> You are then volunteering for something? 2009/3/31 > First off. I would not call it a "janitors project" since that may offend > some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would say > you have that backwards. There is nothing more effective in evolving > FreeSwitch than good documentation which helps further development and is an > important part of "customer service." Good customer service is then a part > of "sales and marketing." Much more often than not, It's sales and marketing > that is more important to making something a "real product" than > engineering. "Build it and they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org < > freeswitch-users at lists.freeswitch.org>; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's going to > keep growing as the word spreads. The core development team of Anthony, > Mike, and Brian are very appreciative of the community's help and > involvement in the project. Simply put: the community is awesome! > > Some have asked how they can help. Most of us are not software developers, > but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To > this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, takes > out the trash, replaces the toilet paper, etc. These are valuable services > that we sometimes take for granted. However, I think we can all appreciate > that the FreeSWITCH project would be better served if the developers could > focus on writing code, fixing bugs, etc. and not on the easier, > not-quite-as-important janitorial tasks. To that end we are inviting all who > wish to volunteer to please visit the above wiki page and check out some of > the projects listed so far. Email me off list if you'd like to volunteer to > help. I'm maintaining a list of "janitors" and what they are helping with. > If you have ideas for other janitor projects then by all means email them to > me and we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > New Low Prices on Dell Laptops - Starting at $399 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/6936e797/attachment-0002.html From mszlazak at aol.com Wed Apr 1 00:12:33 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 03:12:33 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <2021f8b20904010000i11611a1ewa4577d6d97ce9ba8@mail.gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <2021f8b20904010000i11611a1ewa4577d6d97ce9ba8@mail.gmail.com> Message-ID: <8CB80B05D0C3E71-7D8-38C@webmail-mf17.sysops.aol.com> I just did, and it was suggestion. -----Original Message----- From: Larry Edelstein To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 12:00 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects You are then volunteering for something? 2009/3/31 First off. I would not call it a "janitors project" since that may offend some. A second problem is your notion that documentation is "not-quite-as-important" a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of "customer service." Good customer service is then a part of "sales and marketing." Much more often than not, It's sales and marketing that is more important to making something a "real product"? than engineering. "Build it and they will come" almost never works. Anyway, I think you need a new name for this project. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org ; freeswitch-dev at lists.freeswitch.org Sent: Tue, 31 Mar 2009 5:10 pm Subject: [Freeswitch-users] Call For Help: Janitor Projects Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a "janitor projects" wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say "janitor" projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of "janitors" and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009!? http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org New Low Prices on Dell Laptops - Starting at $399 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/9da061d6/attachment-0002.html From dujinfang at gmail.com Wed Apr 1 00:33:37 2009 From: dujinfang at gmail.com (seven) Date: Wed, 1 Apr 2009 15:33:37 +0800 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> Message-ID: <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> Agree, I think the author better to document the code first. For a simple example: if you add a new param or channel variable, at least should add an item to the wiki, so others knows there is a new variable and try that add add detailed explanation or experience further. On Apr 1, 2009, at 2:21 PM, mszlazak at aol.com wrote: > First off. I would not call it a "janitors project" since that may > offend some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many > would say you have that backwards. There is nothing more effective > in evolving FreeSwitch than good documentation which helps further > development and is an important part of "customer service." Good > customer service is then a part of "sales and marketing." Much more > often than not, It's sales and marketing that is more important to > making something a "real product" than engineering. "Build it and > they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org >; freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's > going to keep growing as the word spreads. The core development team > of Anthony, Mike, and Brian are very appreciative of the community's > help and involvement in the project. Simply put: the community is > awesome! > > Some have asked how they can help. Most of us are not software > developers, but that doesn't mean we can't help to grow the > FreeSWITCH ecosystem. To this end I've started a "janitor projects" > wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, > takes out the trash, replaces the toilet paper, etc. These are > valuable services that we sometimes take for granted. However, I > think we can all appreciate that the FreeSWITCH project would be > better served if the developers could focus on writing code, fixing > bugs, etc. and not on the easier, not-quite-as-important janitorial > tasks. To that end we are inviting all who wish to volunteer to > please visit the above wiki page and check out some of the projects > listed so far. Email me off list if you'd like to volunteer to help. > I'm maintaining a list of "janitors" and what they are helping with. > If you have ideas for other janitor projects then by all means email > them to me and we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > New Low Prices on Dell Laptops - Starting at $399 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/3a79ef33/attachment-0002.html From raul at etellicom.com Wed Apr 1 01:29:56 2009 From: raul at etellicom.com (Raul Fragoso) Date: Wed, 01 Apr 2009 05:29:56 -0300 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> Message-ID: <1238574596.18630.64.camel@raul-laptop> Pardon my honesty, but I think you are the one who is getting this backwards. Firstly, I fail to see why a call for help with organizing and cleaning up the project documentation would offend someone by simply having "janitor" as the name. Have you ever heard the term "gatekeeper" before ? Would it offend you ? Think again. Secondly, FreeSWITCH is an open-source project, so forget the 'marketing & sales' crap in the context of documentation. The success of the project, which is growing incredibly fast, is built upon the collaboration of the community as a whole, and it's common sense that sharing the project tasks is a major necessary step to keep it going, just like a janitor is of primordial importance to keep an office building organized and clean. Last but not the least, I agree entirely with the fact that the core developers should be doing what they do it best, and that is, of course, development. I see this call for help request as an effective way of keeping them developing new features and improving the current functionality of FreeSWITCH while sharing the burden of documentation and organization. That's fair and sounds very logical to me. If you join the FreeSWITCH IRC channel and hang in there for a bit you will understand what I mean, most of the time these guys are busy responding to user questions or analyzing use cases that could be easily solved by checking a more organized documentation, and this is what Michael's request is all about. Regards, Raul On Wed, 2009-04-01 at 02:21 -0400, mszlazak at aol.com wrote: > First off. I would not call it a "janitors project" since that may > offend some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would > say you have that backwards. There is nothing more effective in > evolving FreeSwitch than good documentation which helps further > development and is an important part of "customer service." Good > customer service is then a part of "sales and marketing." Much more > often than not, It's sales and marketing that is more important to > making something a "real product" than engineering. "Build it and > they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > ; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's > going to keep growing as the word spreads. The core development team > of Anthony, Mike, and Brian are very appreciative of the community's > help and involvement in the project. Simply put: the community is > awesome! > > Some have asked how they can help. Most of us are not software > developers, but that doesn't mean we can't help to grow the FreeSWITCH > ecosystem. To this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, > takes out the trash, replaces the toilet paper, etc. These are > valuable services that we sometimes take for granted. However, I think > we can all appreciate that the FreeSWITCH project would be better > served if the developers could focus on writing code, fixing bugs, > etc. and not on the easier, not-quite-as-important janitorial tasks. > To that end we are inviting all who wish to volunteer to please visit > the above wiki page and check out some of the projects listed so far. > Email me off list if you'd like to volunteer to help. I'm maintaining > a list of "janitors" and what they are helping with. If you have ideas > for other janitor projects then by all means email them to me and > we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ______________________________________________________________________ > New Low Prices on Dell Laptops - Starting at $399 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 01:42:47 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 03:42:47 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> Message-ID: <1180B0C2-2791-4523-B0CD-1EB13213059F@freeswitch.org> What do you recommend calling it then? I wouldn't be offended by it ... and I can't think of any reason it would offend someone because it describes the task at hand. As far as documentation vs code... without the code there would be ZERO need for any documentation. The code is the hardest part to make sure it functions bug free. Developers are great at writing code but not the best at writing documentation, me included. It's the perfect place for anyone that wants to help out! I welcome anyone and everyone to the project in hopes that community members will help out! We have various IRC channels... #freeswitch, #freeswitch-dev, #freeswitch-docs and #freeswitch-social so join irc.freenode.net and get involved because you never know how it might change your life for the better! ;) /b Positive anything is better than negative thinking. On Apr 1, 2009, at 1:21 AM, mszlazak at aol.com wrote: > First off. I would not call it a "janitors project" since that may > offend some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many > would say you have that backwards. There is nothing more effective > in evolving FreeSwitch than good documentation which helps further > development and is an important part of "customer service." Good > customer service is then a part of "sales and marketing." Much more > often than not, It's sales and marketing that is more important to > making something a "real product" than engineering. "Build it and > they will come" almost never works. > > Anyway, I think you need a new name for this project. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/7d5dfda9/attachment-0002.html From jason at jasonjgw.net Wed Apr 1 03:00:58 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Apr 2009 21:00:58 +1100 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> Message-ID: <20090401100058.GA15830@jdc.jasonjgw.net> seven wrote: > Agree, I think the author better to document the code first. Well, actually... it's already done. It's called API documentation, and consists of specially written comments in the code. This is not user-level documentation, however; it exists to help programmers who want to write applications or FreeSWITCH modules, or to participate in the development effort. Keep in mind also that this is a free software/open-source project; the developers are free to decide how best to spend their time. Personally, I would rather that they spend as much of the time as they wish writing and maintaining code. I've read enough of the code in FreeSWITCH to appreciate its high quality and the soundness of the design. It should also be remembered that the source code is the ultimate documentation, and everyone is free to look at it and to document (in their preferred natural language) what they find out. From dujinfang at gmail.com Wed Apr 1 03:45:28 2009 From: dujinfang at gmail.com (seven) Date: Wed, 1 Apr 2009 18:45:28 +0800 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <20090401100058.GA15830@jdc.jasonjgw.net> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> <20090401100058.GA15830@jdc.jasonjgw.net> Message-ID: <7FA73899-C2E5-4B69-B750-0E4700B8E26C@gmail.com> I know that. And I'd like to read code. Developers written great code and also plenty of comments(which is documentation) in code. However, there are sth. don't need to comment in code but should be available on wiki. E.g. I followed the svn commit log, and found sip_auth_username and sip_auth_password added, so I documented to the wiki. On Apr 1, 2009, at 6:00 PM, Jason White wrote: > seven wrote: >> Agree, I think the author better to document the code first. > > Well, actually... it's already done. It's called API documentation, > and > consists of specially written comments in the code. > > This is not user-level documentation, however; it exists to help > programmers > who want to write applications or FreeSWITCH modules, or to > participate in the > development effort. > > Keep in mind also that this is a free software/open-source project; > the > developers are free to decide how best to spend their time. I agree with you, whether of not document to wiki is up to the developers. But I just think it would be better(or more easier) if we(or others) can find all (including all the newest) params or features in wiki so we can try it and add document more on wiki. > > > Personally, I would rather that they spend as much of the time as > they wish > writing and maintaining code. > > I've read enough of the code in FreeSWITCH to appreciate its high > quality and > the soundness of the design. > > It should also be remembered that the source code is the ultimate > documentation, and everyone is free to look at it and to document > (in their > preferred natural language) what they find out. > > So do I. I'd like following the svn commit log to see what's new in there. But not all of us like to or have the time to read source code. Perhaps that's why we are here to help documenting.... > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Wed Apr 1 04:41:06 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 01 Apr 2009 13:41:06 +0200 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> Message-ID: <49D352D2.3070303@gmx.net> Hello Brian, I tried this (on trunk 12862), but still the same behaviour. It does not aks for a PIN. Neither when transfering directly to the conference nor by transfering to the dialplan extension where conference is handled. Best regards Peter Brian West schrieb: > Update again to svn trunk... btw 1.0.4 pre3 is out on > files.freeswitch.org > > /b > > On Mar 30, 2009, at 6:44 PM, Dan Le wrote: > >> I get similar behavior as Peter when trying to enter a locked >> conference. >> >> If I am just dialing from a phone to a conference (on a dialplan), it >> will properly lock me out. But if I do an originate command >> (originate sofia/internal/1001 &conference(3000)), it will drop me >> into the conference, even though it is suppose to be locked. >> >> I am using the released 1.0.3 tag. >> > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 lewisppp at gmail.com Wed Apr 1 03:41:44 2009 From: lewisppp at gmail.com (Lewis Liu) Date: Wed, 1 Apr 2009 18:41:44 +0800 Subject: [Freeswitch-users] Compiler error for Windows XP (SP2) Message-ID: <814e59990904010341y61b920c2h24bb8a50c8ae2f44@mail.gmail.com> We download FreeSWITCH from SVN Trunk and want to build it on MS Visual Studio 2008 with platform. But we got one error message when we build it. FreeSWITCH\libs\win32\sofia\debug\libsofia_sip_ua_static.lib is built fail. So many files are lost, such as mod_sofia.dll..... Could you help me me for this, Please?? Whether something is lost in MS Visual Studio 2008 ?? Thanks a lot!! Lewis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/b79932de/attachment-0002.html From anthony.minessale at gmail.com Wed Apr 1 06:19:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 08:19:54 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> Message-ID: <191c3a030904010619i21ddd8fj81b020340907eb27@mail.gmail.com> have a look. http://www.google.com/search?q=janitor+project The phrase has already been coined. If you look closely we have 2 different perspectives in this thread. mszlazak is seeking more of the higher level user documentation, the holy grail magic documentation that is like the hitchhikers guide to the galaxy or harry potter's marauder's map can tune into what you need to know or what you don't understand and magically adjusts. This is normal, we have a lot of users like that. The majority of users will treat us like they are buying the software from us and impose their expectations on us. It's helpful to us, it lets us see things from their perspective. Seven is looking it at more from a developer's perspective, he's actually willing to take the time to add things to the wiki and he wants to understand how the code works. This is a good thing too, there are far less people of this type in our community but they are crucial. Core developers document by explaining what they are doing to people like Seven or by putting a reminder in the commit notes which are later translated into the CHANGELOG for the releases. Michael, the author of this thread has added countless pages of documentation to the wiki this way. It's easy to say the author should document everything. There is close to 300,000 lines of code in just the src directory in the FreeSWITCH tree (that is all code we wrote not counting any of the depends libs or any other form of pre-existing code). I personally wrote the majority of that code so, I really appricate it when the communiuty gives me a few minutes to take a break while they document it. The best people to document the high level fuctionality is not the author btw. It's the first few people who use it. Most likely they are developing a product from it and they intend to profit from it in one way or another and its a fair tradeoff to have the section of functionality explained to them in exchange from wikifying it from their perspective. The perspective of the author will be dry and mechinacal where that first-time-user version of the documentation will make much more sense to future readers. When it comes to the low level documentation, the C functions, we also need someone to help us with that if they feel there is not enough. We write code, we know how it works. If other people cannot figure out how it works, they will ask us and in the end it will be doucmented. About 5% or less of people in the community even have to look in the code for the core. The whole point of the FreeSWITCH design is to push everything up to scripts, remote connections and dialplan logic to let people concentrate on good ideas instead of the evil logic necessary to properly engineer a telephony engine. So I recommend anybody interested starts out making sure there is ample documentation for the embedded and external API for lua, js, perl, python, ESL etc. Then anybody who really likes C code can start with the module API layer and then dig deeper into the core code and learn how it works and if the documentation is not enough, add some, we appriciate any help we can get. 2009/4/1 > First off. I would not call it a "janitors project" since that may offend > some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would say > you have that backwards. There is nothing more effective in evolving > FreeSwitch than good documentation which helps further development and is an > important part of "customer service." Good customer service is then a part > of "sales and marketing." Much more often than not, It's sales and marketing > that is more important to making something a "real product" than > engineering. "Build it and they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org < > freeswitch-users at lists.freeswitch.org>; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's going to > keep growing as the word spreads. The core development team of Anthony, > Mike, and Brian are very appreciative of the community's help and > involvement in the project. Simply put: the community is awesome! > > Some have asked how they can help. Most of us are not software developers, > but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To > this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, takes > out the trash, replaces the toilet paper, etc. These are valuable services > that we sometimes take for granted. However, I think we can all appreciate > that the FreeSWITCH project would be better served if the developers could > focus on writing code, fixing bugs, etc. and not on the easier, > not-quite-as-important janitorial tasks. To that end we are inviting all who > wish to volunteer to please visit the above wiki page and check out some of > the projects listed so far. Email me off list if you'd like to volunteer to > help. I'm maintaining a list of "janitors" and what they are helping with. > If you have ideas for other janitor projects then by all means email them to > me and we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > New Low Prices on Dell Laptops - Starting at $399 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f8bc7440/attachment-0002.html From mike at jerris.com Wed Apr 1 06:55:10 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 1 Apr 2009 09:55:10 -0400 Subject: [Freeswitch-users] Compiler error for Windows XP (SP2) In-Reply-To: <814e59990904010341y61b920c2h24bb8a50c8ae2f44@mail.gmail.com> References: <814e59990904010341y61b920c2h24bb8a50c8ae2f44@mail.gmail.com> Message-ID: <711C4390-ED0C-4A06-9AE8-652B24D0C776@jerris.com> If you try to build just the sofia library, what are the first few warnings and errors you get? Mike On Apr 1, 2009, at 6:41 AM, Lewis Liu wrote: > We download FreeSWITCH from SVN Trunk and want to build it on MS > Visual Studio 2008 with platform. > But we got one error message when we build it. > FreeSWITCH\libs\win32\sofia\debug\libsofia_sip_ua_static.lib is > built fail. > So many files are lost, such as mod_sofia.dll..... > Could you help me me for this, Please?? > Whether something is lost in MS Visual Studio 2008 ?? > Thanks a lot!! > Lewis From intralanman at freeswitch.org Wed Apr 1 06:59:15 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 01 Apr 2009 09:59:15 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <7FA73899-C2E5-4B69-B750-0E4700B8E26C@gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> <20090401100058.GA15830@jdc.jasonjgw.net> <7FA73899-C2E5-4B69-B750-0E4700B8E26C@gmail.com> Message-ID: <49D37333.5080701@freeswitch.org> seven wrote: > I know that. And I'd like to read code. Developers written great code > and also plenty of comments(which is documentation) in code. However, > there are sth. don't need to comment in code but should be available > on wiki. E.g. I followed the svn commit log, and found > sip_auth_username and sip_auth_password added, so I documented to the > wiki. > That's the right attitude to have... now if there were more people doing that and less people complaining like little school girls, we could actually reach the next level in Open-Sourcetopia. -Ray From anthony.minessale at gmail.com Wed Apr 1 07:30:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 09:30:06 -0500 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <49D352D2.3070303@gmx.net> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> <49D352D2.3070303@gmx.net> Message-ID: <191c3a030904010730h24946e50vf2ca7b2936f776d1@mail.gmail.com> pin checks and lock checks are both intentionally skipped on outbound calls transferred back to the conference. The idea is if you purposely placed an outbound call that was intended to land in the conference you would not want to do so only to tell them it's locked. I added a patch to trunk so you can override this with a variable originate {conference_enforce_security=true}sofia/internal/1001 &conference(3000) the same var can be used on inbound calls for the opposite effect On Wed, Apr 1, 2009 at 6:41 AM, Peter P GMX wrote: > Hello Brian, > > I tried this (on trunk 12862), but still the same behaviour. It does not > aks for a PIN. Neither when transfering directly to the conference nor > by transfering to the dialplan extension where conference is handled. > > Best regards > Peter > > > > Brian West schrieb: > > Update again to svn trunk... btw 1.0.4 pre3 is out on > > files.freeswitch.org > > > > /b > > > > On Mar 30, 2009, at 6:44 PM, Dan Le wrote: > > > >> I get similar behavior as Peter when trying to enter a locked > >> conference. > >> > >> If I am just dialing from a phone to a conference (on a dialplan), it > >> will properly lock me out. But if I do an originate command > >> (originate sofia/internal/1001 &conference(3000)), it will drop me > >> into the conference, even though it is suppose to be locked. > >> > >> I am using the released 1.0.3 tag. > >> > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/82d95c15/attachment-0002.html From jmesquita at gmail.com Wed Apr 1 07:36:36 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 1 Apr 2009 11:36:36 -0300 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <49D37333.5080701@freeswitch.org> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> <20090401100058.GA15830@jdc.jasonjgw.net> <7FA73899-C2E5-4B69-B750-0E4700B8E26C@gmail.com> <49D37333.5080701@freeswitch.org> Message-ID: <2C42C47B-AD14-433F-B80F-9446E87D44F7@gmail.com> I am sorry, but I really have to comment this one. Why the fuck do we need to have sooo much politics on an open source project? Janitor, non-janitor, developer, non-developer, girl or boy, we are all trying to get this thing better, aren't we? So leave your fucking ego out of the question and get your ass doing something that will actually get this project somewhere like we all instead of trying to get yourself called something. You want the president title? Get it and start working. Tony is the master dude in this place because, like he said, he wrote most of the 300,000 line of code. That simple. The title "core developers team" (sounds great, doesn't it?) are because .... they do CORE! Wanna be called core developer, DO CORE! Anyway, my suggestion is, want something done? DO IT. Don't know how? Study! Don't want to know how ... buy Avaya or whatever. They will charge for your laziness. Sorry for the bad language. Mesquita On Apr 1, 2009, at 10:59 AM, Raymond Chandler wrote: > seven wrote: >> I know that. And I'd like to read code. Developers written great code >> and also plenty of comments(which is documentation) in code. However, >> there are sth. don't need to comment in code but should be available >> on wiki. E.g. I followed the svn commit log, and found >> sip_auth_username and sip_auth_password added, so I documented to the >> wiki. >> > That's the right attitude to have... now if there were more people > doing > that and less people complaining like little school girls, we could > actually reach the next level in Open-Sourcetopia. > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Apr 1 07:37:45 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 1 Apr 2009 07:37:45 -0700 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <49D37333.5080701@freeswitch.org> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <365EA298-BF80-4407-8F1F-9B8CE70D4F5A@gmail.com> <20090401100058.GA15830@jdc.jasonjgw.net> <7FA73899-C2E5-4B69-B750-0E4700B8E26C@gmail.com> <49D37333.5080701@freeswitch.org> Message-ID: <37962F11-AAF0-42C5-97BA-C12A72194DA5@freeswitch.org> On Apr 1, 2009, at 6:59 AM, Raymond Chandler wrote: > seven wrote: >> I know that. And I'd like to read code. Developers written great code >> and also plenty of comments(which is documentation) in code. However, >> there are sth. don't need to comment in code but should be available >> on wiki. E.g. I followed the svn commit log, and found >> sip_auth_username and sip_auth_password added, so I documented to the >> wiki. >> > That's the right attitude to have... now if there were more people > doing > that and less people complaining like little school girls, we could > actually reach the next level in Open-Sourcetopia. > > -Ray First off, thank you all for your thoughts. This thread has yielded far more passion than I had hoped for. I consider that a good thing. It's okay for us to share differing opinions. Enthusiastic disagreements are better than ambivalence. :) Secondly, I just want to say that I like the term "janitor" because of its connotation. A janitor is someone who puts forth effort doing honorable work. A literal janitor is trusted with the keys to the office and leaves the workplace in a better condition than when he or she arrived. Another word for janitor is custodian. Please view the word in this positive light: a trusted worker whose contributions are valued by all. Thirdly, I want to thank people for stepping up. I've already received several private emails from volunteers. Please feel free to inundate my inbox! Lastly, I'd just like to thank Anthony, Brian, and Mike for devoting so much time and energy to FreeSWITCH. They've created a wonderful product, and they've also invested a lot of time answering my questions and those of others. I feel it's the least I can do to try and get that knowledge codified into a usable format so others can benefit also. Thanks again for your thoughts, ideas, and opinions. Keep them coming! We may just yet reach Open Sourcetopia. -MC From anthony.minessale at gmail.com Wed Apr 1 08:31:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 10:31:09 -0500 Subject: [Freeswitch-users] Another FreeSWITCH First! Message-ID: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> The FreeSWITCH team is excited to announce that FreeSWITCH is the first telephony application to support the new SIP 4.1 protocol specification. Unlike its predecessors, SIP 4.1 has been created with the collaboration of both the jabber foundation and the IETF. With this match made in heaven, one can now encapsulate an xml representation of a sip message, which in turn can encapsulate a standard SIP 2.0 message making it possible to do more than ever before. Other exciting features include: *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT with ease. *) Full circle presence: endpoints must subscribe to each character in the printable ASCII range that may be used to indicate presence and the server will send an xml notification to the client for each character that is enabled whenever a call takes place which in turn can be used to build a SIP 4.1 FYI packet that can be sent to all the neighboring SIP devices so they may send themselves a NOTIFY telling them that the light should blink if the same packet happens to be sent from a neighbor. Then when the neighbor wants to send a presence packet it establishes a dialog with the Third Party Presence Agent TPPA and leaves the message there. Then it sends the server a PRESENCE packet, which is then, relayed to the subscribers with the TPPA id so all the subscribers can connect to the TPPA server to make the little light blink. *) Retirement of SDP: SDP is deprecated in favor of a list of URL?s describing the desired codec. The UA can then request this URL and get the full details of the media requirements. The media port is negotiated through trial and error where the calling UA asks the called UA if the port it has guessed randomly is correct via direct TCP connection and an exchange of XML PORT MARKUP LANGUGE XPML INVITE bob at alice.com SIP 4.1 Content-type: sip-xml-encapsulated -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/d6ddf3b8/attachment-0002.html From anthony.minessale at gmail.com Wed Apr 1 08:51:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 10:51:07 -0500 Subject: [Freeswitch-users] Long Lost Comments Surface, Now We Know... Message-ID: <191c3a030904010851m4c416ab8qfb41c04d731a8490@mail.gmail.com> In one of the most suprising events in current technology history in this modern era, the long lost comments to many of the now-adopted internet RFC's have finally surfaced. Aparently the mail server was misconfigured at "The Internet Society" and most of the comments were redirected to the local lost-and-found box on the server whey they sat for decades. In a suprising twist, it appears that RFC 2543, the predacessor to 3261 regarding the session initnation protocol had several critisisms that went unanswered. A few examples are included below. Some other comments in regards to RFC2833 (RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals) could not be published due to the graphic nature of the content. From: alice at anywhere.com to:comments at tis.org Subject: RFC 2543 Ahem, who gave you permission to use my name in your document? Also, how did you find out about me and Bob? Thanks to you it's all over the net >=0 From: jwlt at columbia.edu to:comments at tis.org Subject: RFC 2543 Are you guys sure about this? We were pretty drunk last night. I didn't think you would actually go through with it! lol I was just kiddding! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/57e729f8/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Apr 1 09:07:36 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 1 Apr 2009 17:07:36 +0100 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> Message-ID: Well you almost had me there, but SIP over SMTP? That was too much. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 01 April 2009 16:31 To: Freeswitch-users Subject: [Freeswitch-users] Another FreeSWITCH First! The FreeSWITCH team is excited to announce that FreeSWITCH is the first telephony application to support the new SIP 4.1 protocol specification. Unlike its predecessors, SIP 4.1 has been created with the collaboration of both the jabber foundation and the IETF. With this match made in heaven, one can now encapsulate an xml representation of a sip message, which in turn can encapsulate a standard SIP 2.0 message making it possible to do more than ever before. Other exciting features include: *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT with ease. *) Full circle presence: endpoints must subscribe to each character in the printable ASCII range that may be used to indicate presence and the server will send an xml notification to the client for each character that is enabled whenever a call takes place which in turn can be used to build a SIP 4.1 FYI packet that can be sent to all the neighboring SIP devices so they may send themselves a NOTIFY telling them that the light should blink if the same packet happens to be sent from a neighbor. Then when the neighbor wants to send a presence packet it establishes a dialog with the Third Party Presence Agent TPPA and leaves the message there. Then it sends the server a PRESENCE packet, which is then, relayed to the subscribers with the TPPA id so all the subscribers can connect to the TPPA server to make the little light blink. *) Retirement of SDP: SDP is deprecated in favor of a list of URL's describing the desired codec. The UA can then request this URL and get the full details of the media requirements. The media port is negotiated through trial and error where the calling UA asks the called UA if the port it has guessed randomly is correct via direct TCP connection and an exchange of XML PORT MARKUP LANGUGE XPML INVITE bob at alice.com SIP 4.1 Content-type: sip-xml-encapsulated -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/8c815602/attachment-0002.html From brian at freeswitch.org Wed Apr 1 09:15:24 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 11:15:24 -0500 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> Message-ID: You know you could write a transport plugin for Sofia that would do SIP over SMTP :P /b On Apr 1, 2009, at 11:07 AM, Nik Middleton wrote: > Well you almost had me there, but SIP over SMTP? That was too much. > > Regards, > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/ddfe7158/attachment-0002.html From msc at freeswitch.org Wed Apr 1 09:34:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 09:34:54 -0700 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <191c3a030904010730h24946e50vf2ca7b2936f776d1@mail.gmail.com> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> <49D352D2.3070303@gmx.net> <191c3a030904010730h24946e50vf2ca7b2936f776d1@mail.gmail.com> Message-ID: <87f2f3b90904010934i567f9a15k13a33b2e125fd69c@mail.gmail.com> 2009/4/1 Anthony Minessale > pin checks and lock checks are both intentionally skipped on outbound calls > transferred back to the conference. > The idea is if you purposely placed an outbound call that was intended to > land in the conference > you would not want to do so only to tell them it's locked. > > I added a patch to trunk so you can override this with a variable > > originate {conference_enforce_security=true}sofia/internal/1001 > &conference(3000) > > the same var can be used on inbound calls for the opposite effect > > > > > > FYI this is now in the wiki: http://wiki.freeswitch.org/wiki/Channel_Variables#conference_enforce_security -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f2127449/attachment-0002.html From msc at freeswitch.org Wed Apr 1 09:49:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 09:49:47 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre3 Now Available Message-ID: <87f2f3b90904010949x4d3fc2f8ia9feda2581956f68@mail.gmail.com> The FreeSWITCH team would like to let everyone know that the latest version is available. More information can be found here: http://www.freeswitch.org/node/172 By all means download, upgrade, test, and report back! Your feedback helps us make FreeSWITCH even better! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/deb90856/attachment-0002.html From edpimentl at gmail.com Wed Apr 1 09:53:15 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 1 Apr 2009 12:53:15 -0400 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> Message-ID: <9dc4a1670904010953x4f1f6742h3fc1f355af23baa4@mail.gmail.com> LOVE!!!!! Now we can create Twitter-Voip apps.... Best regards, -E CEO and Founder Gpro.ws edpimentl [SKype | GoogleTalk | Twitter ] http://Twitter.com/edpimentl http://AskTwitR.com (Real Time Twitter Search & Reputation Management) http://TwiTR.Me (Cross Social Network Messaging Bus) http://TweetOnTV.net (Private Label Social TV Platform) http://TwebEX.com (Twitter Based Online Web Conference Platform) http://TwitrShare.com (Send Picture and Message to Tweet Contacts) http://Twookups.com (Twitter Matching Service) http://TweetUp.ws (Twitter based MeetUp service) 2009/4/1 Anthony Minessale > The FreeSWITCH team is excited to announce that FreeSWITCH is the first > telephony application to support the new SIP 4.1 protocol specification. > > Unlike its predecessors, SIP 4.1 has been created with the collaboration of > both the jabber foundation and the IETF. With this match made in heaven, > one can now encapsulate an xml representation of a sip message, which in > turn can encapsulate a standard SIP 2.0 message making it possible to do > more than ever before. > Other exciting features include: > > *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT with > ease. > > *) Full circle presence: endpoints must subscribe to each character in the > printable ASCII range that may be used to indicate presence and the server > will send an xml notification to the client for each character that is > enabled whenever a call takes place which in turn can be used to build a SIP > 4.1 FYI packet that can be sent to all the neighboring SIP devices so they > may send themselves a NOTIFY telling them that the light should blink if the > same packet happens to be sent from a neighbor. Then when the neighbor > wants to send a presence packet it establishes a dialog with the Third Party > Presence Agent TPPA and leaves the message there. Then it sends the server > a PRESENCE packet, which is then, relayed to the subscribers with the TPPA > id so all the subscribers can connect to the TPPA server to make the little > light blink. > > *) Retirement of SDP: SDP is deprecated in favor of a list of URL?s > describing the desired codec. The UA can then request this URL and get the > full details of the media requirements. The media port is negotiated > through trial and error where the calling UA asks the called UA if the port > it has guessed randomly is correct via direct TCP connection and an exchange > of XML PORT MARKUP LANGUGE XPML > > INVITE bob at alice.com SIP 4.1 > Content-type: sip-xml-encapsulated > > > > > To: bob at alice.com > From: alice at bob.com > Subject: SIP Rocks > ]]> > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/8f43290d/attachment-0002.html From brian at freeswitch.org Wed Apr 1 09:53:38 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 11:53:38 -0500 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre3 Now Available In-Reply-To: <87f2f3b90904010949x4d3fc2f8ia9feda2581956f68@mail.gmail.com> References: <87f2f3b90904010949x4d3fc2f8ia9feda2581956f68@mail.gmail.com> Message-ID: Which btw this is NOT an april fools joke! Its really 1.0.4 pre3 ;) /b On Apr 1, 2009, at 11:49 AM, Michael Collins wrote: > The FreeSWITCH team would like to let everyone know that the latest > version is available. More information can be found here: > http://www.freeswitch.org/node/172 > > By all means download, upgrade, test, and report back! Your feedback > helps us make FreeSWITCH even better! > -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 Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/23c6155a/attachment-0002.html From mszlazak at aol.com Wed Apr 1 10:24:45 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 13:24:45 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <1238574596.18630.64.camel@raul-laptop> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> Message-ID: <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> Pardon me, but you speak only for yourself. I think Janitor is not an appropriate word. Second, 'marketing and sales' does not only mean making money. It also means 'selling' someone on the idea of trying something and effectively spreading the word. Third, the original developers can spend most of their time developing because they're the creators so they know very well what's going on with the code and don't need good documentation. Others need good documentation to effectively work with FS or do development. Currently the documentation is scattered, assumes to much and is outdated/incorrected. Also, there is a problem with not getting the "creators" involved with documentation since someone doing the documentation will have to ask them what's what. The "creators" never will be totally out of the loop nor should they be. This doesn't apply only here in this context but other similar ones as well. Keeping? "creators" from inteact with "customers" is one big reason so many start-ups fail. -----Original Message----- From: Raul Fragoso To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 1:29 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Pardon my honesty, but I think you are the one who is getting this backwards. Firstly, I fail to see why a call for help with organizing and cleaning up the project documentation would offend someone by simply having "janitor" as the name. Have you ever heard the term "gatekeeper" before ? Would it offend you ? Think again. Secondly, FreeSWITCH is an open-source project, so forget the 'marketing & sales' crap in the context of documentation. The success of the project, which is growing incredibly fast, is built upon the collaboration of the community as a whole, and it's common sense that sharing the project tasks is a major necessary step to keep it going, just like a janitor is of primordial importance to keep an office building organized and clean. Last but not the least, I agree entirely with the fact that the core developers should be doing what they do it best, and that is, of course, development. I see this call for help request as an effective way of keeping them developing new features and improving the current functionality of FreeSWITCH while sharing the burden of documentation and organization. That's fair and sounds very logical to me. If you join the FreeSWITCH IRC channel and hang in there for a bit you will understand what I mean, most of the time these guys are busy responding to user questions or analyzing use cases that could be easily solved by checking a more organized documentation, and this is what Michael's request is all about. Regards, Raul On Wed, 2009-04-01 at 02:21 -0400, mszlazak at aol.com wrote: > First off. I would not call it a "janitors project" since that may > offend some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would > say you have that backwards. There is nothing more effective in > evolving FreeSwitch than good documentation which helps further > development and is an important part of "customer service." Good > customer service is then a part of "sales and marketing." Much more > often than not, It's sales and marketing that is more important to > making something a "real product" than engineering. "Build it and > they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > ; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's > going to keep growing as the word spreads. The core development team > of Anthony, Mike, and Brian are very appreciative of the community's > help and involvement in the project. Simply put: the community is > awesome! > > Some have asked how they can help. Most of us are not software > developers, but that doesn't mean we can't help to grow the FreeSWITCH > ecosystem. To this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, > takes out the trash, replaces the toilet paper, etc. These are > valuable services that we sometimes take for granted. However, I think > we can all appreciate that the FreeSWITCH project would be better > served if the developers could focus on writing code, fixing bugs, > etc. and not on the easier, not-quite-as-important janitorial tasks. > To that end we are inviting all who wish to volunteer to please visit > the above wiki page and check out some of the projects listed so far. > Email me off list if you'd like to volunteer to help. I'm maintaining > a list of "janitors" and what they are helping with. If you have ideas > for other janitor projects then by all means email them to me and > we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ______________________________________________________________________ > New Low Prices on Dell Laptops - Starting at $399 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/325e0e6c/attachment-0002.html From brian at freeswitch.org Wed Apr 1 10:39:44 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 12:39:44 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> Message-ID: <8534C3EE-DFE4-4743-8975-EAD9B2BCE1D8@freeswitch.org> Are you referring to PocketSphinx here? /b On Apr 1, 2009, at 12:24 PM, mszlazak at aol.com wrote: > Currently the documentation is scattered, assumes to much and is > outdated/incorrected. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/85fed388/attachment-0002.html From mszlazak at aol.com Wed Apr 1 10:45:48 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 13:45:48 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <1180B0C2-2791-4523-B0CD-1EB13213059F@freeswitch.org> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1180B0C2-2791-4523-B0CD-1EB13213059F@freeswitch.org> Message-ID: <8CB8108D36771BE-458-31AD@webmail-dh09.sysops.aol.com> Call it what it is like "The Documentation Project" or something similar. Sure, if there was no code there is no FS but I didn't say the code is not important. I was taking a sales/marketing versus engineering analogy to this and only said that many would find it less important than good documentation if you are looking to get people to use FS and/or evolve the code. So as long as the creators of FS are willing to work to some extent on the documentation with a documentor, when one is needed, then this should work out. The creators have a very good understanding of FS which the documentor may not. On the other hand, the documentor doesn't have the creators background baggage which makes things seem obvious to the creator but isn't to users or even other developers. The creators and documentors working together will hopefully make the FS documentation accurate, not to presumptuous and easy to use. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 1:42 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects What do you recommend calling it then? ?I wouldn't be offended by it ... and I can't think of any reason it would offend someone because it describes the task at hand. ?As far as documentation vs code... without the code there would be ZERO need for any documentation. ?The code is the hardest part to make sure it functions bug free. ?Developers are great at writing code but not the best at writing documentation, me included. ?It's the perfect place for anyone that wants to help out! ?I welcome anyone and everyone to the project in hopes that community members will help out! ? We have various IRC channels... #freeswitch, #freeswitch-dev, #freeswitch-docs and #freeswitch-social so join irc.freenode.net and get involved because you never know how it might change your life for the better! ;) /b Positive anything is better than negative thinking. On Apr 1, 2009, at 1:21 AM, mszlazak at aol.com wrote: First off. I would not call it a "janitors project" since that may offend some. A second problem is your notion that documentation is "not-quite-as-important" a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of "customer service." Good customer service is then a part of "sales and marketing." Much more often than not, It's sales and marketing that is more important to making something a "real product"? than engineering. "Build it and they will come" almost never works. Anyway, I think you need a new name for this project. Brian West brian at freeswitch.org -- Meet us a ClueCon! ?http://www.cluecon.com = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/18d238f6/attachment-0002.html From brian at freeswitch.org Wed Apr 1 10:52:47 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 12:52:47 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB8108D36771BE-458-31AD@webmail-dh09.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1180B0C2-2791-4523-B0CD-1EB13213059F@freeswitch.org> <8CB8108D36771BE-458-31AD@webmail-dh09.sysops.aol.com> Message-ID: On Apr 1, 2009, at 12:45 PM, mszlazak at aol.com wrote: > Call it what it is like "The Documentation Project" or something > similar. Because its MORE than Documentation! So that name is silly! > > Sure, if there was no code there is no FS but I didn't say the code > is not important. I was taking a sales/marketing versus engineering > analogy to this and only said that many would find it less important > than good documentation if you are looking to get people to use FS > and/or evolve the code. So as long as the creators of FS are willing > to work to some extent on the documentation with a documentor, when > one is needed, then this should work out. The creators have a very > good understanding of FS which the documentor may not. On the other > hand, the documentor doesn't have the creators background baggage > which makes things seem obvious to the creator but isn't to users or > even other developers. The creators and documentors working together > will hopefully make the FS documentation accurate, not to > presumptuous and easy to use. Well if people join IRC... ask questions we do answer them... so if people don't understand something all they have to do is ask we won't bite. > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/8b735633/attachment-0002.html From mszlazak at aol.com Wed Apr 1 10:56:02 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 13:56:02 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <191c3a030904010619i21ddd8fj81b020340907eb27@mail.gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <191c3a030904010619i21ddd8fj81b020340907eb27@mail.gmail.com> Message-ID: <8CB810A41678FCC-458-3259@webmail-dh09.sysops.aol.com> "The holy grail magic documentation that is like the hitchhikers guide to the galaxy or harry potter's marauder's map can tune into what you need to know or what you don't understand and magically adjusts."! Maybe your projecting or exaggerating but I didn't say anything like that. However, the important point was "we have a lot of users like that." Enough said. -----Original Message----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 6:19 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects have a look. http://www.google.com/search?q=janitor+project The phrase has already been coined. If you look closely we have 2 different perspectives in this thread. mszlazak is seeking more of the higher level user documentation, the holy grail magic documentation that is like the hitchhikers guide to the galaxy or harry potter's marauder's map can tune into what you need to know or what you don't understand and magically adjusts.? This is normal,?.? The majority of users will treat us like they are buying the software from us and impose their expectations on us.? It's helpful to us, it lets us see things from their perspective. Seven is looking it at more from a developer's perspective, he's actually willing to take the time to add things to the wiki and he wants to understand how the code works.? This is a good thing too, there are far less people of this type in our community but they are crucial.? Core developers document by explaining what they are doing to people like Seven or by putting a reminder in the commit notes which are later translated into the CHANGELOG for the releases.? Michael, the author of this thread has added countless pages of documentation to the wiki this way.? It's easy to say the author should document everything.? There is close to 300,000 lines of code in just the src directory in the FreeSWITCH tree (that is all code we wrote not counting any of the depends libs or any other form of pre-existing code).? I personally wrote the majority of that code so, I really appricate it when the communiuty gives me a few minutes to take a break while they document it.? The best people to document the high level fuctionality? is not the author btw.? It's the first few people who use it.? Most likely they are developing a product from it and they intend to profit from it in one way or another and its a fair tradeoff to have the section of functionality explained to them in exchange from wikifying it from their perspective.? The perspective of the author will be dry and mechinacal where that first-time-user version of the documentation will make much more sense to future readers.? When it comes to the low level documentation, the C functions, we also need someone to help us with that if they feel there is not enough.? We write code, we know how it works.? If other people cannot figure out how it works, they will ask us and in the end it will be doucmented.? About 5% or less of people in the community even have to look in the code for the core.? The whole point of the FreeSWITCH design is to push everything up to scripts, remote connections and dialplan logic to let people concentrate on good ideas instead of the evil logic necessary to properly engineer a telephony engine.? So I recommend anybody interested starts out making sure there is ample documentation for the embedded and external API for lua, js, perl, python, ESL etc.? Then anybody who really likes C code can start with the module API layer and then dig deeper into the core code and learn how it works and if the documentation is not enough, add some, we appriciate any help we can get. ? 2009/4/1 First off. I would not call it a "janitors project" since that may offend some. A second problem is your notion that documentation is "not-quite-as-important" a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of "customer service." Good customer service is then a part of "sales and marketing." Much more often than not, It's sales and marketing that is more important to making something a "real product"? than engineering. "Build it and they will come" almost never works. Anyway, I think you need a new name for this project. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org ; freeswitch-dev at lists.freeswitch.org Sent: Tue, 31 Mar 2009 5:10 pm Subject: [Freeswitch-users] Call For Help: Janitor Projects Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a "janitor projects" wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say "janitor" projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of "janitors" and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009!? http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org New Low Prices on Dell Laptops - Starting at $399 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/c9887216/attachment-0002.html From mszlazak at aol.com Wed Apr 1 10:56:37 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 13:56:37 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8534C3EE-DFE4-4743-8975-EAD9B2BCE1D8@freeswitch.org> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com><1238574596.18630.64.camel@raul-laptop><8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <8534C3EE-DFE4-4743-8975-EAD9B2BCE1D8@freeswitch.org> Message-ID: <8CB810A565E5D2A-458-3264@webmail-dh09.sysops.aol.com> nope -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 10:39 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Are you referring to PocketSphinx here?? /b On Apr 1, 2009, at 12:24 PM, mszlazak at aol.com wrote: ?Currently the documentation is scattered, assumes to much and is outdated/incorrected. Brian West brian at freeswitch.org -- Meet us a ClueCon! ?http://www.cluecon.com = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/8e0aee51/attachment-0002.html From peter at cindyandpeter.com Wed Apr 1 10:58:57 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Wed, 1 Apr 2009 13:58:57 -0400 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> Message-ID: <02be01c9b2f3$87d77050$978650f0$@com> Excellent stuff Anthony! J SIP over SMTP could actually be useful in a push-to-talk type of scenario. Put the voice packets in an attachment. A slight delay, perhaps, but nicely encapsulated in a totally standard protocol. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: Wednesday, April 01, 2009 12:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Another FreeSWITCH First! Well you almost had me there, but SIP over SMTP? That was too much. Regards, _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 01 April 2009 16:31 To: Freeswitch-users Subject: [Freeswitch-users] Another FreeSWITCH First! The FreeSWITCH team is excited to announce that FreeSWITCH is the first telephony application to support the new SIP 4.1 protocol specification. Unlike its predecessors, SIP 4.1 has been created with the collaboration of both the jabber foundation and the IETF. With this match made in heaven, one can now encapsulate an xml representation of a sip message, which in turn can encapsulate a standard SIP 2.0 message making it possible to do more than ever before. Other exciting features include: *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT with ease. *) Full circle presence: endpoints must subscribe to each character in the printable ASCII range that may be used to indicate presence and the server will send an xml notification to the client for each character that is enabled whenever a call takes place which in turn can be used to build a SIP 4.1 FYI packet that can be sent to all the neighboring SIP devices so they may send themselves a NOTIFY telling them that the light should blink if the same packet happens to be sent from a neighbor. Then when the neighbor wants to send a presence packet it establishes a dialog with the Third Party Presence Agent TPPA and leaves the message there. Then it sends the server a PRESENCE packet, which is then, relayed to the subscribers with the TPPA id so all the subscribers can connect to the TPPA server to make the little light blink. *) Retirement of SDP: SDP is deprecated in favor of a list of URL's describing the desired codec. The UA can then request this URL and get the full details of the media requirements. The media port is negotiated through trial and error where the calling UA asks the called UA if the port it has guessed randomly is correct via direct TCP connection and an exchange of XML PORT MARKUP LANGUGE XPML INVITE bob at alice.com SIP 4.1 Content-type: sip-xml-encapsulated -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/ef4291f4/attachment-0002.html From mszlazak at aol.com Wed Apr 1 11:02:22 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 14:02:22 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com><1180B0C2-2791-4523-B0CD-1EB13213059F@freeswitch.org><8CB8108D36771BE-458-31AD@webmail-dh09.sysops.aol.com> Message-ID: <8CB810B23ADBD5C-458-32D3@webmail-dh09.sysops.aol.com> Excellent! The core developers/creators should stay active in the documentation process. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 10:52 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects On Apr 1, 2009, at 12:45 PM, mszlazak at aol.com wrote: Call it what it is like "The Documentation Project" or something similar. Because its MORE than Documentation! ?So that name is silly! Sure, if there was no code there is no FS but I didn't say the code is not important.?I was taking a sales/marketing versus engineering analogy to this and?only said that many would find it less important than good documentation if you are looking to get people to use FS and/or evolve the code. So as long as the creators of FS are willing to work to some extent on the documentation with a documentor, when one is needed, then this should work out. The creators have a very good understanding of FS which the documentor may not. On the other hand, the documentor doesn't have the creators background baggage which makes things seem obvious to the creator but isn't to users or even other developers. The creators and documentors working together will hopefully make the FS documentation accurate, not to presumptuous and easy to use. Well if people join IRC... ask questions we do answer them... so if people don't understand something all they have to do is ask we won't bite. Brian West brian at freeswitch.org -- Meet us a ClueCon! ?http://www.cluecon.com = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/83b50667/attachment-0002.html From msc at freeswitch.org Wed Apr 1 11:18:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 11:18:26 -0700 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> Message-ID: <87f2f3b90904011118o28e4196bn50068353fa5ae8ea@mail.gmail.com> 2009/4/1 > Pardon me, but you speak only for yourself. I think Janitor is not an > appropriate word. > I *like* janitors. I *respect* janitors. They are *honorable* and * hard-working*. In short, we need janitors - people who are willing to roll up their sleeves and get work done. Let's agree to disagree on this word. If you don't like the word janitor then I will respect your viewpoint. Use the word "custodian" instead. However, the developers and all the core "power-users" have no qualms with the use of the word janitor. They will be called janitor projects; this point is not up for discussion. Let's all move on. As to your other points: yes, the core developers are involved in the documentation. They don't micromanage, but they give direction. When something is wrong they point it out. When there is a need, they make it known. When they get asked a lot of questions on a specific topic they tell me there's a need for documentation on the subject. Also, we have a number of users who are watching the mailing list and IRC channel who take it upon themselves to document the various nuggets of wisdom that get passed around in the threads. And I do my best to do same-day documentation when Anthony adds a new channel variable or new functionality to a module. As for documentation being outdated/scattered/incomplete/: Many of these observations are valid. There are serious needs - a lot of stuff needs cleaning up. (Which, ironically, is what *janitors* do very well.) However, let me make this point very clear: general statements like "the docs are out of date" are all but worthless. What we need are specific statements, like "I tried to follow the wiki instructions on pocketsphinx but I think they might be outdated or incorrect. May I discuss it with someone in the know?" All such specific comments are welcome. They can be sent to me personally, to this list, or on IRC. FYI, we do have a channel specifically for documentation discussion: #freeswitch-docs. Please join that channel to discuss this subject in real-time. All that being said, here's the bottom line: If you're willing to help then please do so. If you aren't sure where to start then contact me off list and we'll discuss it. If you have have positive feedback then please publish it publicly. If you have negative feedback, criticism, complaints, etc. then please send it to me in private. I've got my coveralls, my mop, and my bucket. Who's with me? :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/53b0ac30/attachment-0002.html From raul at etellicom.com Wed Apr 1 11:21:40 2009 From: raul at etellicom.com (Raul Fragoso) Date: Wed, 01 Apr 2009 15:21:40 -0300 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: <02be01c9b2f3$87d77050$978650f0$@com> References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> <02be01c9b2f3$87d77050$978650f0$@com> Message-ID: <1238610100.10390.9.camel@raul-laptop> Agreed 100% ! That means we are all closer on taking 'mail-agents' to the holy-grail level of voice communications ! I wonder if SIP 4.1 UAS will also handle MX records ? That would be awesome ! I can't wait until we see something like mod_audio_spammer in FreeSWITCH, so those lovely marketing workers can give voice to their so much acclaimed phallic products. Regards, Raul On Wed, 2009-04-01 at 13:58 -0400, Peter J. Zandvoort wrote: > Excellent stuff Anthony! J > > > > SIP over SMTP could actually be useful in a push-to-talk type of > scenario. Put the voice packets in an attachment. A slight delay, > perhaps, but nicely encapsulated in a totally standard protocol. > > > > > > From:freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Nik Middleton > Sent: Wednesday, April 01, 2009 12:08 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Another FreeSWITCH First! > > > > > Well you almost had me there, but SIP over SMTP? That was too much. > > > > Regards, > > > > > ______________________________________________________________________ > From:freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony Minessale > Sent: 01 April 2009 16:31 > To: Freeswitch-users > Subject: [Freeswitch-users] Another FreeSWITCH First! > > > > > The FreeSWITCH team is excited to announce that FreeSWITCH is the > first telephony application to support the new SIP 4.1 protocol > specification. > > Unlike its predecessors, SIP 4.1 has been created with the > collaboration of both the jabber foundation and the IETF. With this > match made in heaven, one can now encapsulate an xml representation of > a sip message, which in turn can encapsulate a standard SIP 2.0 > message making it possible to do more than ever before. > Other exciting features include: > > *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT > with ease. > > *) Full circle presence: endpoints must subscribe to each character in > the printable ASCII range that may be used to indicate presence and > the server will send an xml notification to the client for each > character that is enabled whenever a call takes place which in turn > can be used to build a SIP 4.1 FYI packet that can be sent to all the > neighboring SIP devices so they may send themselves a NOTIFY telling > them that the light should blink if the same packet happens to be sent > from a neighbor. Then when the neighbor wants to send a presence > packet it establishes a dialog with the Third Party Presence Agent > TPPA and leaves the message there. Then it sends the server a > PRESENCE packet, which is then, relayed to the subscribers with the > TPPA id so all the subscribers can connect to the TPPA server to make > the little light blink. > > *) Retirement of SDP: SDP is deprecated in favor of a list of URL?s > describing the desired codec. The UA can then request this URL and > get the full details of the media requirements. The media port is > negotiated through trial and error where the calling UA asks the > called UA if the port it has guessed randomly is correct via direct > TCP connection and an exchange of XML PORT MARKUP LANGUGE XPML > > INVITE bob at alice.com SIP 4.1 > Content-type: sip-xml-encapsulated > > > > > To: bob at alice.com > From: alice at bob.com > Subject: SIP Rocks > ]]> > > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 11:23:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 13:23:48 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> Message-ID: <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> Did you follow the link I posted? http://www.google.com/search?q=janitor+project The linux kernel calls it the same thing and so do all the other project that come up in that search. Would you prefer "Custodial Engineering projects" I tried to be nice but you continue to perpetuate this thread. Another term you may not be familiar with is when someone who is outnumbered starts trying to get the last word on a mailing list or forum, they're called "trolls" Exactly how much have you contributed to this project other than complaints? You initially contacted us at our consulting address, where we then called you on the phone and helped you for 2 hours for free even though we know your goal is to develop a product from FreeSWITCH and most people in your position offer to pay us for our time. (make as many products as you want, that's why we made FreeSWITCH so good for you, but, usually if you want *that much* help you have to pay for it) You started using modules that were just written at the time you came around on a platform on which the module only was compiling for a week, give us a break..... We have all helped you on the list and documented things *for you* on several dozen occasions. I don't want anything in return but for you to please stop commenting on this thread. This is not a mob rule project, I will make the decisions for it when I see fit and when I seek the input of others, I ask for it and when I don't want any input I do whatever I want. It's a perk of running your own project. I personally don't care what Collins calls it, janitor project or whatever, at least he is show initiative and getting people involved. 2009/4/1 > Pardon me, but you speak only for yourself. I think Janitor is not an > appropriate word. > > Second, 'marketing and sales' does not only mean making money. It also > means 'selling' someone on the idea of trying something and effectively > spreading the word. > > Third, the original developers can spend most of their time developing > because they're the creators so they know very well what's going on with the > code and don't need good documentation. Others need good documentation to > effectively work with FS or do development. Currently the documentation is > scattered, assumes to much and is outdated/incorrected. Also, there is a > problem with not getting the "creators" involved with documentation since > someone doing the documentation will have to ask them what's what. The > "creators" never will be totally out of the loop nor should they be. This > doesn't apply only here in this context but other similar ones as well. > Keeping "creators" from inteact with "customers" is one big reason so many > start-ups fail. > > -----Original Message----- > From: Raul Fragoso > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, 1 Apr 2009 1:29 am > Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects > > Pardon my honesty, but I think you are the one who is getting this > > backwards. > > > Firstly, I fail to see why a call for help with organizing and cleaning > > up the project documentation would offend someone by simply having > > "janitor" as the name. Have you ever heard the term "gatekeeper" > > before ? Would it offend you ? Think again. > > > Secondly, FreeSWITCH is an open-source project, so forget the 'marketing > > & sales' crap in the context of documentation. The success of the > > project, which is growing incredibly fast, is built upon the > > collaboration of the community as a whole, and it's common sense that > > sharing the project tasks is a major necessary step to keep it going, > > just like a janitor is of primordial importance to keep an office > > building organized and clean. > > > Last but not the least, I agree entirely with the fact that the core > > developers should be doing what they do it best, and that is, of course, > > development. I see this call for help request as an effective way of > > keeping them developing new features and improving the current > > functionality of FreeSWITCH while sharing the burden of documentation > > and organization. That's fair and sounds very logical to me. If you join > > the FreeSWITCH IRC channel and hang in there for a bit you will > > understand what I mean, most of the time these guys are busy responding > > to user questions or analyzing use cases that could be easily solved by > > checking a more organized documentation, and this is what Michael's > > request is all about. > > > Regards, > > > Raul > > > On Wed, 2009-04-01 at 02:21 -0400, mszlazak at aol.com wrote: > > > First off. I would not call it a "janitors project" since that may > > > offend some. A second problem is your notion that documentation is > > > "not-quite-as-important" a task as writing code. I'm think many would > > > say you have that backwards. There is nothing more effective in > > > evolving FreeSwitch than good documentation which helps further > > > development and is an important part of "customer service." Good > > > customer service is then a part of "sales and marketing." Much more > > > often than not, It's sales and marketing that is more important to > > > making something a "real product" than engineering. "Build it and > > > they will come" almost never works. > > > > > > Anyway, I think you need a new name for this project. > > > > > > > > > > > > > > > > > > -----Original Message----- > > > From: Michael Collins > > > To: freeswitch-users at lists.freeswitch.org > > > ; > > > freeswitch-dev at lists.freeswitch.org > > > Sent: Tue, 31 Mar 2009 5:10 pm > > > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > > > > > Dear FreeSWITCH Community: > > > > > > As you know, FreeSWITCH has been growing leaps and bounds and it's > > > going to keep growing as the word spreads. The core development team > > > of Anthony, Mike, and Brian are very appreciative of the community's > > > help and involvement in the project. Simply put: the community is > > > awesome! > > > > > > Some have asked how they can help. Most of us are not software > > > developers, but that doesn't mean we can't help to grow the FreeSWITCH > > > ecosystem. To this end I've started a "janitor projects" wiki page: > > > > > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > > > > > We say "janitor" projects because they are things that help keep the > > > project clean and organized, just like the janitor cleans an office, > > > takes out the trash, replaces the toilet paper, etc. These are > > > valuable services that we sometimes take for granted. However, I think > > > we can all appreciate that the FreeSWITCH project would be better > > > served if the developers could focus on writing code, fixing bugs, > > > etc. and not on the easier, not-quite-as-important janitorial tasks. > > > To that end we are inviting all who wish to volunteer to please visit > > > the above wiki page and check out some of the projects listed so far. > > > Email me off list if you'd like to volunteer to help. I'm maintaining > > > a list of "janitors" and what they are helping with. If you have ideas > > > for other janitor projects then by all means email them to me and > > > we'll discuss them. > > > > > > Thanks again for being such a great community! > > > > > > -Michael S Collins > > > IRC: mercutioviz > > > > > > See you at ClueCon 2009! http://www.cluecon.com > > > > > > _______________________________________________ > > > > > > Freeswitch-users mailing list > > > > > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > > > > ______________________________________________________________________ > > > New Low Prices on Dell Laptops - Starting at $399 > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > New Low Prices on Dell Laptops - Starting at $399 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/3dfd737f/attachment-0002.html From grevenx at me.com Wed Apr 1 11:23:39 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Wed, 01 Apr 2009 20:23:39 +0200 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <87f2f3b90904010934i567f9a15k13a33b2e125fd69c@mail.gmail.com> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> <49D352D2.3070303@gmx.net> <191c3a030904010730h24946e50vf2ca7b2936f776d1@mail.gmail.com> <87f2f3b90904010934i567f9a15k13a33b2e125fd69c@mail.gmail.com> Message-ID: <95833A23-7543-4E88-8445-93FAE9CD00AE@me.com> You're one very fine janitor Michael! On the topic of the Janitor Project, this is how it should be. Devs give user feature => user documents new feature/behaviour. Even Andr? On 1. april. 2009, at 18.34, Michael Collins wrote: > > > 2009/4/1 Anthony Minessale > pin checks and lock checks are both intentionally skipped on > outbound calls transferred back to the conference. > The idea is if you purposely placed an outbound call that was > intended to land in the conference > you would not want to do so only to tell them it's locked. > > I added a patch to trunk so you can override this with a variable > > originate {conference_enforce_security=true}sofia/internal/1001 > &conference(3000) > > the same var can be used on inbound calls for the opposite effect > > > > > > > FYI this is now in the wiki: > http://wiki.freeswitch.org/wiki/Channel_Variables#conference_enforce_security > > -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/20090401/ef6a3779/attachment-0002.html From intralanman at freeswitch.org Wed Apr 1 11:29:13 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 01 Apr 2009 14:29:13 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> Message-ID: <49D3B279.6040901@freeswitch.org> mszlazak at aol.com wrote: > Pardon me, but you speak only for yourself. I think Janitor is not an > appropriate word. you're welcome to your opinions, no matter how wrong they are > > Second, 'marketing and sales' does not only mean making money. It also > means 'selling' someone on the idea of trying something and > effectively spreading the word. > we don't try to sell anyone on the project... we'll tell you the pros and cons, you decide if the software meets your needs or not. > Third, the original developers can spend most of their time developing > because they're the creators so they know very well what's going on > with the code and don't need good documentation. Others need good > documentation to effectively work with FS or do development. Currently > the documentation is scattered, assumes to much and is > outdated/incorrected. maybe you could fix some of that since you seem to be very enlightened to its shortcomings? although, that might offend your delicate psyche since you'd basically be a "janitor" then. > Also, there is a problem with not getting the "creators" involved with > documentation since someone doing the documentation will have to ask > them what's what. The "creators" never will be totally out of the loop > nor should they be. This doesn't apply only here in this context but > other similar ones as well. Keeping "creators" from inteact with > "customers" is one big reason so many start-ups fail. > hmmm, maybe you're right... maybe the whole idea of hierarchy is entirely wrong. i guess we could expect tony to document his own code... while we're at it, let's suggest that microsoft has Bill Gates write documentation for windows and answer tech support calls, right? cus i mean, obviously everyone who writes code should obviously do everything else too, right? but i guess that doesn't work the other direction... cus if you don't know how to code, then you just can't code... its as simple as that. so now we have effectively halved (or better) the development activities of FreeSWITCH so there's less to document, but that's ok, because now there's plenty of people using it and not contributing anything back... and that's what open-source is really all about, right? btw, i'm just curious if you're an employee of a commercial entity that feels threatened by FreeSWITCH... what better way to decrease productivity than to split hairs over something so stupid as the name of an effort (janitor projects, in this case) that you're not going to take part in anyway. if i may ask, have you done anything constructive for the community at all? all i've seen of you from the mailing lists is non-constructive criticisms. not that we don't appreciate your trolling... its very entertaining to see how narrow-minded some people are. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/03b6bb8b/attachment-0002.html From carlos.talbot at gmail.com Wed Apr 1 11:40:58 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 1 Apr 2009 13:40:58 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt Message-ID: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> Is there an interest in running FreeSWITCH on OpenWRT? I recently managed to compile and run a version for a MIPs based router, the Planex MZK-W04NU. This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP space. I was planning to setup a wiki page on compiling and configuring. regards, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/ddb2b8a5/attachment-0002.html From msc at freeswitch.org Wed Apr 1 11:43:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 11:43:03 -0700 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <95833A23-7543-4E88-8445-93FAE9CD00AE@me.com> References: <49CBEA8D.4050901@gmx.net> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> <49D352D2.3070303@gmx.net> <191c3a030904010730h24946e50vf2ca7b2936f776d1@mail.gmail.com> <87f2f3b90904010934i567f9a15k13a33b2e125fd69c@mail.gmail.com> <95833A23-7543-4E88-8445-93FAE9CD00AE@me.com> Message-ID: <87f2f3b90904011143j465c462er2b44b173a2ba412e@mail.gmail.com> 2009/4/1 Even Andr? Fiskvik > You're one very fine janitor Michael! > How DARE you call me a janitor! :) > On the topic of the Janitor Project, this is how it should be. > Devs give user feature => user documents new feature/behaviour. > Thanks. This is totally reasonable. Power users and newbies both can add to the documentation. If anyone has questions about how to help or would like some pointers then by all means contact me off list. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/ea2a010f/attachment-0002.html From timr at asteriasgi.com Wed Apr 1 11:46:39 2009 From: timr at asteriasgi.com (Tim Ringenbach) Date: Wed, 1 Apr 2009 13:46:39 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> Message-ID: <49D3B68F.7010806@asteriasgi.com> Anthony Minessale wrote: > Did you follow the link I posted? > http://www.google.com/search?q=janitor+project > > The linux kernel calls it the same thing and so do all the other > project that come up in that search. > > Would you prefer "Custodial Engineering projects" > It definitely is the commonly used term for that sort of thing. But I would tend to agree that I wouldn't expect people to get excited about volunteering to be a janitor. Any idea how successful those projects are at attracting volunteers? Sadly, I don't have a better suggestion. But no matter how much Michael says he loves janitors, to me a janitor is someone who has to clean up other people's crap (figuratively and sometimes literally). And I can see how that could fail to attract as many volunteers as the "Freeswitch Happy, Rich, and Well Endowed people" project might. > I tried to be nice but you continue to perpetuate this thread. > > Another term you may not be familiar with is when someone who is > outnumbered starts trying to > get the last word on a mailing list or forum, they're called "trolls" I always thought trolls had to be trying to really be considered a troll. Like if I were to post to this list trying to convince you all to give up on freeswitch and join the asterisk project, while knowing full well the history, and just trying to get a rise out of you. --Tim From msc at freeswitch.org Wed Apr 1 11:50:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 11:50:09 -0700 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> Message-ID: <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> 2009/4/1 Carlos Talbot > > Is there an interest in running FreeSWITCH on OpenWRT? I recently managed > to compile and run a version for a MIPs based router, the Planex MZK-W04NU. > This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support > (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks > online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP > space. > Just curious - is there a use case for doing this, other than the hobbyist who does it because it's cool? > > I was planning to setup a wiki page on compiling and configuring. Please do. We like to see all the different places and ways that people use FreeSWITCH. -MC > > > regards, > > Carlos > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/8503d3b2/attachment-0002.html From msc at freeswitch.org Wed Apr 1 11:54:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 11:54:43 -0700 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <49D3B68F.7010806@asteriasgi.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> <49D3B68F.7010806@asteriasgi.com> Message-ID: <87f2f3b90904011154l4c07279eg555c9574f168fb1a@mail.gmail.com> On Wed, Apr 1, 2009 at 11:46 AM, Tim Ringenbach wrote: > Anthony Minessale wrote: > > Did you follow the link I posted? > > http://www.google.com/search?q=janitor+project > > > > The linux kernel calls it the same thing and so do all the other > > project that come up in that search. > > > > Would you prefer "Custodial Engineering projects" > > > It definitely is the commonly used term for that sort of thing. But I > would tend to agree that I wouldn't expect people to get excited about > volunteering to be a janitor. Any idea how successful those projects are > at attracting volunteers? > > Sadly, I don't have a better suggestion. But no matter how much Michael > says he loves janitors, to me a janitor is someone who has to clean up > other people's crap (figuratively and sometimes literally). And I can > see how that could fail to attract as many volunteers as the "Freeswitch > Happy, Rich, and Well Endowed people" project might. Like I said: Grab a mop and bucket or get outta the way! It's time to take out the trash. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/acd46ee4/attachment-0002.html From rupa at rupa.com Wed Apr 1 11:56:18 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 1 Apr 2009 13:56:18 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> Message-ID: 2009/4/1 Michael Collins > 2009/4/1 Carlos Talbot > >> >> Is there an interest in running FreeSWITCH on OpenWRT? I recently managed >> to compile and run a version for a MIPs based router, the Planex MZK-W04NU. >> This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support >> (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks >> online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP >> space. >> > > Just curious - is there a use case for doing this, other than the hobbyist > who does it because it's cool? > I could see using it as a standalone product for (very) small businesses or as a home gateway+phone. Guess the biggest issue would be lack of reasonable local storage for voicemail. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/347e0dd1/attachment-0002.html From anthony.minessale at gmail.com Wed Apr 1 12:02:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 14:02:11 -0500 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <49D3B68F.7010806@asteriasgi.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop> <8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> <49D3B68F.7010806@asteriasgi.com> Message-ID: <191c3a030904011202u74238814x78ee46c7674c994@mail.gmail.com> how about: "WALL-E projects" maybe Steve J will give us permission. On Wed, Apr 1, 2009 at 1:46 PM, Tim Ringenbach wrote: > Anthony Minessale wrote: > > Did you follow the link I posted? > > http://www.google.com/search?q=janitor+project > > > > The linux kernel calls it the same thing and so do all the other > > project that come up in that search. > > > > Would you prefer "Custodial Engineering projects" > > > It definitely is the commonly used term for that sort of thing. But I > would tend to agree that I wouldn't expect people to get excited about > volunteering to be a janitor. Any idea how successful those projects are > at attracting volunteers? > > Sadly, I don't have a better suggestion. But no matter how much Michael > says he loves janitors, to me a janitor is someone who has to clean up > other people's crap (figuratively and sometimes literally). And I can > see how that could fail to attract as many volunteers as the "Freeswitch > Happy, Rich, and Well Endowed people" project might. > > I tried to be nice but you continue to perpetuate this thread. > > > > Another term you may not be familiar with is when someone who is > > outnumbered starts trying to > > get the last word on a mailing list or forum, they're called "trolls" > I always thought trolls had to be trying to really be considered a > troll. Like if I were to post to this list trying to convince you all to > give up on freeswitch and join the asterisk project, while knowing full > well the history, and just trying to get a rise out of you. > > --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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/c5643291/attachment-0002.html From carlos.talbot at gmail.com Wed Apr 1 12:02:06 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 1 Apr 2009 14:02:06 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> Message-ID: <5800526b0904011202s3f98757xc3df0327c2432ed1@mail.gmail.com> Until I figure out how much of a load it can handle for now it's just an experiment. :) I was motivated by two factors: - Kristin had ported FreeSWITCH to Astlinux, another uClib embedded environment. This sparked my interest in getting it to work on OpenWRT - Asterisk has been running on OpenWRT for a while so I wanted to see how difficult it would be to bring in FreeSWITCH. Carlos 2009/4/1 Michael Collins > 2009/4/1 Carlos Talbot > >> >> Is there an interest in running FreeSWITCH on OpenWRT? I recently managed >> to compile and run a version for a MIPs based router, the Planex MZK-W04NU. >> This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support >> (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks >> online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP >> space. >> > > Just curious - is there a use case for doing this, other than the hobbyist > who does it because it's cool? > > >> >> I was planning to setup a wiki page on compiling and configuring. > > > Please do. We like to see all the different places and ways that people use > FreeSWITCH. > > -MC > > >> >> >> regards, >> >> Carlos >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/b81c9443/attachment-0002.html From stevecrozz at gmail.com Wed Apr 1 12:09:34 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 1 Apr 2009 12:09:34 -0700 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <5800526b0904011202s3f98757xc3df0327c2432ed1@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> <5800526b0904011202s3f98757xc3df0327c2432ed1@mail.gmail.com> Message-ID: <11990ade0904011209t133f12b4sc40246976be8972a@mail.gmail.com> Sounds like a fun project. I wouldn't worry too much about the lack of local storage space for voicemail. You can easily mount remote filesystems to increase storage capacity. I've done so using openwrt for my own projects using shfs, nfs, and next I want to try s3fs. --Stephen 2009/4/1 Carlos Talbot > Until I figure out how much of a load it can handle for now it's just an > experiment. :) > > I was motivated by two factors: > > - Kristin had ported FreeSWITCH to Astlinux, another uClib embedded > environment. This sparked my interest in getting it to work on OpenWRT > - Asterisk has been running on OpenWRT for a while so I wanted to see how > difficult it would be to bring in FreeSWITCH. > > Carlos > > 2009/4/1 Michael Collins > >> 2009/4/1 Carlos Talbot >> >> >>> Is there an interest in running FreeSWITCH on OpenWRT? I recently managed >>> to compile and run a version for a MIPs based router, the Planex MZK-W04NU. >>> This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support >>> (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks >>> online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP >>> space. >>> >> >> Just curious - is there a use case for doing this, other than the hobbyist >> who does it because it's cool? >> >> >>> >>> I was planning to setup a wiki page on compiling and configuring. >> >> >> Please do. We like to see all the different places and ways that people >> use FreeSWITCH. >> >> -MC >> >> >>> >>> >>> regards, >>> >>> Carlos >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f4b78d03/attachment-0002.html From cesar.bermudez at gmail.com Wed Apr 1 13:27:27 2009 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Wed, 1 Apr 2009 22:27:27 +0200 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> Message-ID: where can see and buy that router? 2009/4/1 Carlos Talbot > > Is there an interest in running FreeSWITCH on OpenWRT? I recently managed > to compile and run a version for a MIPs based router, the Planex MZK-W04NU. > This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support > (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks > online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP > space. > > I was planning to setup a wiki page on compiling and configuring. > > regards, > > Carlos > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f74c109c/attachment-0002.html From mszlazak at aol.com Wed Apr 1 13:29:31 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 16:29:31 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com><1238574596.18630.64.camel@raul-laptop><8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <191c3a030904011123g463b36ceha413e71f89cef28c@mail.gmail.com> Message-ID: <8CB811FB2C391ED-698-1157@webmail-dd17.sysops.aol.com> You tried to be nice! Give me a break. Maybe try harder next time. -----Original Message----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 11:23 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Did you follow the link I posted? http://www.google.com/search?q=janitor+project The linux kernel calls it the same thing and so do all the other project that come up in that search. Would you prefer "Custodial Engineering projects" I tried to be nice but you continue to perpetuate this thread. Another term you may not be familiar with is when someone who is outnumbered starts trying to get the last word on a mailing list or forum, they're called "trolls" Exactly how much have you contributed to this project other than complaints? You initially contacted us at our consulting address, where we then called you on the phone and helped you for 2 hours for free even though we know your goal is to develop a product from FreeSWITCH and most people in your position offer to pay us for our time.? (make as many products as you want, that's why we made FreeSWITCH so good for you, but, usually if you want *that much* help you have to pay for it) You started using modules that were just written at the time you came around on a platform on which the module only was compiling for a week, give us a break..... We have all helped you on the list and documented things *for you* on several dozen occasions. I don't want anything in return but for you to please stop commenting on this thread. This is not a mob rule project, I will make the decisions for it when I see fit and when I seek the input of others, I ask for it and when I don't want any input I do whatever I want.? It's a perk of running your own project.? I personally don't care what Collins calls it, janitor project or whatever, at least he is show initiative and getting people involved. 2009/4/1 Pardon me, but you speak only for yourself. I think Janitor is not an appropriate word. Second, 'marketing and sales' does not only mean making money. It also means 'selling' someone on the idea of trying something and effectively spreading the word. Third, the original developers can spend most of their time developing because they're the creators so they know very well what's going on with the code and don't need good documentation. Others need good documentation to effectively work with FS or do development. Currently the documentation is scattered, assumes to much and is outdated/incorrected. Also, there is a problem with not getting the "creators" involved with documentation since someone doing the documentation will have to ask them what's what. The "creators" never will be totally out of the loop nor should they be. This doesn't apply only here in this context but other similar ones as well. Keeping? "creators" from inteact with "customers" is one big reason so many start-ups fail. -----Original Message----- From: Raul Fragoso To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 1:29 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Pardon my honesty, but I think you are the one who is getting this backwards. Firstly, I fail to see why a call for help with organizing and cleaning up the project documentation would offend someone by simply having "janitor" as the name. Have you ever heard the term "gatekeeper" before ? Would it offend you ? Think again. Secondly, FreeSWITCH is an open-source project, so forget the 'marketing & sales' crap in the context of documentation. The success of the project, which is growing incredibly fast, is built upon the collaboration of the community as a whole, and it's common sense that sharing the project tasks is a major necessary step to keep it going, just like a janitor is of primordial importance to keep an office building organized and clean. Last but not the least, I agree entirely with the fact that the core developers should be doing what they do it best, and that is, of course, development. I see this call for help request as an effective way of keeping them developing new features and improving the current functionality of FreeSWITCH while sharing the burden of documentation and organization. That's fair and sounds very logical to me. If you join the FreeSWITCH IRC channel and hang in there for a bit you will understand what I mean, most of the time these guys are busy responding to user questions or analyzing use cases that could be easily solved by checking a more organized documentation, and this is what Michael's request is all about. Regards, Raul On Wed, 2009-04-01 at 02:21 -0400, mszlazak at aol.com wrote: > First off. I would not call it a "janitors project" since that may > offend some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would > say you have that backwards. There is nothing more effective in > evolving FreeSwitch than good documentation which helps further > development and is an important part of "customer service." Good > customer service is then a part of "sales and marketing." Much more > often than not, It's sales and marketing that is more important to > making something a "real product" than engineering. "Build it and > they will come" almost never works. > > Anyway, I think you need a new name for this project. > > > > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > ; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > > Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's > going to keep growing as the word spreads. The core development team > of Anthony, Mike, and Brian are very appreciative of the community's > help and involvement in the project. Simply put: the community is > awesome! > > Some have asked how they can help. Most of us are not software > developers, but that doesn't mean we can't help to grow the FreeSWITCH > ecosystem. To this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, > takes out the trash, replaces the toilet paper, etc. These are > valuable services that we sometimes take for granted. However, I think > we can all appreciate that the FreeSWITCH project would be better > served if the developers could focus on writing code, fixing bugs, > etc. and not on the easier, not-quite-as-important janitorial tasks. > To that end we are inviting all who wish to volunteer to please visit > the above wiki page and check out some of the projects listed so far. > Email me off list if you'd like to volunteer to help. I'm maintaining > a list of "janitors" and what they are helping with. If you have ideas > for other janitor projects then by all means email them to me and > we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009! http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ______________________________________________________________________ > New Low Prices on Dell Laptops - Starting at $399 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org New Low Prices on Dell Laptops - Starting at $399 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/317c092a/attachment-0002.html From mszlazak at aol.com Wed Apr 1 13:31:01 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 16:31:01 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <49D3B279.6040901@freeswitch.org> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop><8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <49D3B279.6040901@freeswitch.org> Message-ID: <8CB811FE8046E7F-698-1171@webmail-dd17.sysops.aol.com> You missed the point again. But suffer fools to long. -----Original Message----- From: Raymond Chandler To: freeswitch-users at lists.freeswitch.org Sent: Wed, 1 Apr 2009 11:29 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects mszlazak at aol.com wrote: Pardon me, but you speak only for yourself. I think Janitor is not an appropriate word. you're welcome to your opinions, no matter how wrong they are Second, 'marketing and sales' does not only mean making money. It also means 'selling' someone on the idea of trying something and effectively spreading the word. we don't try to sell anyone on the project... we'll tell you the pros and cons, you decide if the software meets your needs or not. Third, the original developers can spend most of their time developing because they're the creators so they know very well what's going on with the code and don't need good documentation. Others need good documentation to effectively work with FS or do development. Currently the documentation is scattered, assumes to much and is outdated/incorrected. maybe you could fix some of that since you seem to be very enlightened to its shortcomings? although, that might offend your delicate psyche since you'd basically be a "janitor" then. Also, there is a problem with not getting the "creators" involved with documentation since someone doing the documentation will have to ask them what's what. The "creators" never will be totally out of the loop nor should they be. This doesn't apply only here in this context but other similar ones as well. Keeping? "creators" from inteact with "customers" is one big reason so many start-ups fail. hmmm, maybe you're right... maybe the whole idea of hierarchy is entirely wrong. i guess we could expect tony to document his own code... while we're at it, let's suggest that microsoft has Bill Gates write documentation for windows and answer tech support calls, right? cus i mean, obviously everyone who writes code should obviously do everything else too, right? but i guess that doesn't work the other direction... cus if you don't know how to code, then you just can't code... its as simple as that. so now we have effectively halved (or better) the development activities of FreeSWITCH so there's less to document, but that's ok, because now there's plenty of people using it and not contributing anything back... and that's what open-source is really all about, right? btw, i'm just curious if you're an employee of a commercial entity that feels threatened by FreeSWITCH... what better way to decrease productivity than to split hairs over something so stupid as the name of an effort (janitor projects, in this case) that you're not going to take part in anyway. if i may ask, have you done anything constructive for the community at all? all i've seen of you from the mailing lists is non-constructive criticisms. not that we don't appreciate your trolling... its very entertaining to see how narrow-minded some people are. -Ray _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/4f8d39c9/attachment-0002.html From valentin.doroga at pronexus.com Wed Apr 1 14:00:28 2009 From: valentin.doroga at pronexus.com (Valentin Doroga) Date: Wed, 1 Apr 2009 17:00:28 -0400 Subject: [Freeswitch-users] Nokia N800 In-Reply-To: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> Message-ID: <20090401210045.LGSN1685.tomts37-srv.bellnexxia.net@toip34-bus.srvr.bell.ca> There are some old binaries at: http://www.freeswitch.org/downloads/n800/ Is there a newer version? Any place with instruction to build? Val. From dave at 3c.co.uk Wed Apr 1 14:00:25 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 1 Apr 2009 14:00:25 -0700 Subject: [Freeswitch-users] Another FreeSWITCH First! In-Reply-To: References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com> Message-ID: Here's a sample SIP/SMTP INVITE (responses omitted for clarity) MAIL FROM: RCPT TO: DATA Call me . --Dave Sent from my iPhone On 1 Apr 2009, at 09:15, Brian West wrote: > You know you could write a transport plugin for Sofia that would do > SIP over SMTP :P > > /b > > On Apr 1, 2009, at 11:07 AM, Nik Middleton wrote: > >> Well you almost had me there, but SIP over SMTP? That was too much. >> >> Regards, >> > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f8fd5191/attachment-0002.html From intralanman at freeswitch.org Wed Apr 1 14:04:36 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 01 Apr 2009 17:04:36 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <8CB811FE8046E7F-698-1171@webmail-dd17.sysops.aol.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com><8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> <1238574596.18630.64.camel@raul-laptop><8CB8105E2F12740-458-3053@webmail-dh09.sysops.aol.com> <49D3B279.6040901@freeswitch.org> <8CB811FE8046E7F-698-1171@webmail-dd17.sysops.aol.com> Message-ID: <49D3D6E4.1060301@freeswitch.org> mszlazak at aol.com wrote: > You missed the point again. But suffer fools to long. No, I think you missed the point... several times. The point that most of us are trying to make is "if you're not going to help, you have no room to talk". Although, I guess your approach works for you. If you're clearly outwitted, resort to name calling. I've seen a couple of people, including myself, ask if you've done anything except complain. I have not, however, seen you reply with anything intelligent or any contributions that you have made. So to try to make the point again. If you're not contributing anything, then leave us all alone. Hopefully, you're not so feeble-minded that you miss it twice in the same email. If you offer up ideas and they are accepted or considered, then you are a contributor. The point at which you offer your ideas and several members of the community, including the most involved, all disagree with you... you become a troll. It would be greatly apprciated by all persons involved if you, and your misguided opinions, would just concede and leave this thread alone. We now return you to the troll-free "Call For Help" -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/68f1613a/attachment-0002.html From brian at freeswitch.org Wed Apr 1 14:11:13 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 16:11:13 -0500 Subject: [Freeswitch-users] Nokia N800 In-Reply-To: <20090401210045.LGSN1685.tomts37-srv.bellnexxia.net@toip34-bus.srvr.bell.ca> References: <20090401210045.LGSN1685.tomts37-srv.bellnexxia.net@toip34-bus.srvr.bell.ca> Message-ID: We haven't updated it recently... You should be able to use scratch box to accomplish it also. On that note please do not hijack threads... you clicked reply, changed the subject and body which causes it to thread your message with the original posters thread. So please in the future click new message and input freeswitch-users at lists.freeswitch.org Thanks, Brian On Apr 1, 2009, at 4:00 PM, Valentin Doroga wrote: > There are some old binaries at: > http://www.freeswitch.org/downloads/n800/ > > Is there a newer version? Any place with instruction to build? > Val. > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/44d85cd7/attachment-0002.html From stormin.normin at hotmail.co.uk Wed Apr 1 14:09:03 2009 From: stormin.normin at hotmail.co.uk (Stromin Normin) Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Message-ID: Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. _________________________________________________________________ Share your photos with Windows Live Photos ? Free. http://clk.atdmt.com/UKM/go/134665338/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/cabdfc34/attachment-0002.html From toofics at gmail.com Wed Apr 1 13:19:26 2009 From: toofics at gmail.com (Victor Toofic) Date: Wed, 01 Apr 2009 14:19:26 -0600 Subject: [Freeswitch-users] Missing CHANNEL_HANGUP event in mod_event_socket Message-ID: <1238617166.3750.93.camel@ktulu> Hi all!! I'm stuck trying to use mod_event_socket in outbound mode. The problem that I'm facing is that while in a incoming call, using "myevents" to monitor for the channel's events.. the event CHANNEL_HANGUP sometimes arrives and sometimes doesn't. I can't figure it out why. The dialplan is: The process that handles the connection does: 1. connect 2. myevents (received: Reply-Text: +OK Events Enabled) 3. sendmsg\n call-command: execute\n execute-app-name: answer (received: Reply-Text: +OK) after this it waits for events and/or for the other party to hangup the call. (The DTMFs are for testing propourses). Sometimes the events that the process receives are: <<"CHANNEL_PARK">> <<"CHANNEL_EXECUTE">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"CHANNEL_EXECUTE">> <<"CHANNEL_ANSWER">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"DTMF">> <<"DTMF">> <<"CHANNEL_HANGUP">> (then it receives the "text/disconnect-notice" and the socket gets closed) and sometimes are: <<"CHANNEL_EXECUTE">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"CHANNEL_EXECUTE">> <<"CHANNEL_ANSWER">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"DTMF">> <<"DTMF">> (then it receives the "text/disconnect-notice" and the socket gets closed) As you can see, even sometimes the first CHANNEL_PARK event doesn't arrive. I'm very concerned about the missing CHANNEL_HANGUP event. In the other hand I was watching the events in a inbound connection to mod_event_socket with "event text all" and in this case there was no problem, all the events arrived as expected. Why in outbound mode some events get lost?? I'm missing something?? I've tried it in two different machines and the results are the same. I'm using FreeSWITCH Version 1.0.3 (exported) on linux. Thnks!! -- Regards.. Victor Toofic From rupa at rupa.com Wed Apr 1 14:13:04 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 1 Apr 2009 16:13:04 -0500 Subject: [Freeswitch-users] new module: mod_memcache Message-ID: Announcing a new module: mod_memcache Up until now one had two choices for storing arbitrary key/value pairs. hash or db. hash is fast, but it is local to the current FreeSWITCH instance. If you run multiple instances of FreeSWITCH then one could use db, an ODBC connection and a centralized database server (eg: postgresql). The choice was between fast but isolated or slow and distributed. memcached (http://www.danga.com/memcached/) is a high-performance, distributed memory object caching system, generic in nature, but intended for use in speeding up dynamic web applications by alleviating database load. Only now you can use it for dynamic phone applications. Check out the wiki page at: http://wiki.freeswitch.org/wiki/Mod_memcache Try this module out and file bug (jira) reports for problems / enhancement requests. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/beefa377/attachment-0002.html From rupa at rupa.com Wed Apr 1 14:15:33 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 1 Apr 2009 16:15:33 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <11990ade0904011209t133f12b4sc40246976be8972a@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> <87f2f3b90904011150n4885ea95re08fdb84b87f867f@mail.gmail.com> <5800526b0904011202s3f98757xc3df0327c2432ed1@mail.gmail.com> <11990ade0904011209t133f12b4sc40246976be8972a@mail.gmail.com> Message-ID: s3fs would be ideal if this is a turnkey solution. still need local storage (flash) for the sqlite databases, but that shouldn't be very hard. 2009/4/1 Stephen Crosby > Sounds like a fun project. I wouldn't worry too much about the lack of > local storage space for voicemail. You can easily mount remote filesystems > to increase storage capacity. I've done so using openwrt for my own projects > using shfs, nfs, and next I want to try s3fs. > > --Stephen > > > 2009/4/1 Carlos Talbot > >> Until I figure out how much of a load it can handle for now it's just an >> experiment. :) >> >> I was motivated by two factors: >> >> - Kristin had ported FreeSWITCH to Astlinux, another uClib embedded >> environment. This sparked my interest in getting it to work on OpenWRT >> - Asterisk has been running on OpenWRT for a while so I wanted to see how >> difficult it would be to bring in FreeSWITCH. >> >> Carlos >> >> 2009/4/1 Michael Collins >> >>> 2009/4/1 Carlos Talbot >>> >>> >>>> Is there an interest in running FreeSWITCH on OpenWRT? I recently >>>> managed to compile and run a version for a MIPs based router, the Planex >>>> MZK-W04NU. This router has 32MB ram, 8MB flash, runs at 400MHz, draft N >>>> support (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 >>>> bucks online. I used a 2GB USB flash drive for the FreeSWITCH directory and >>>> SWAP space. >>>> >>> >>> Just curious - is there a use case for doing this, other than the >>> hobbyist who does it because it's cool? >>> >>> >>>> >>>> I was planning to setup a wiki page on compiling and configuring. >>> >>> >>> Please do. We like to see all the different places and ways that people >>> use FreeSWITCH. >>> >>> -MC >>> >>> >>>> >>>> >>>> regards, >>>> >>>> Carlos >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090401/88e399e4/attachment-0002.html From anthony.minessale at gmail.com Wed Apr 1 14:18:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 16:18:03 -0500 Subject: [Freeswitch-users] Nokia N800 In-Reply-To: <20090401210045.LGSN1685.tomts37-srv.bellnexxia.net@toip34-bus.srvr.bell.ca> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> <20090401210045.LGSN1685.tomts37-srv.bellnexxia.net@toip34-bus.srvr.bell.ca> Message-ID: <191c3a030904011418x7d0b1e0fifb84e9c514dc51fd@mail.gmail.com> we relocated the machine with the build env for that, I'll try to find the time to resurrect it and make a new one. On Wed, Apr 1, 2009 at 4:00 PM, Valentin Doroga < valentin.doroga at pronexus.com> wrote: > There are some old binaries at: > http://www.freeswitch.org/downloads/n800/ > > Is there a newer version? Any place with instruction to build? > Val. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/c0b5cfc9/attachment-0002.html From msc at freeswitch.org Wed Apr 1 14:18:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 14:18:44 -0700 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: <87f2f3b90904011418j371e1363m4ec1def2e9ba2818@mail.gmail.com> 2009/4/1 Stromin Normin > Hi, > > I've been asked to do some testing on Freeswitch by work, we currently use > Asterisk. I'm quite new to telephony so please go easy. > > I have FS setup on a windows box and at the moment I'm testing internal > calls only, when I transfer calls or call extensions everything sounds > great. The problem occurrs when I setup conferencing, people can log in ok > and we can talk, the trouble is as people start to talk a buzzing sound is > heard in the background, once the talking stops the buzzing stops. If the > person goes on mute there is no buzzing. > Out of curiosity, what kind of phones are you using? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/052816ec/attachment-0002.html From anthony.minessale at gmail.com Wed Apr 1 14:22:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 16:22:08 -0500 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: <191c3a030904011422r2f14d05ancae35f1ed3c8f09d@mail.gmail.com> the buzzing is probably a 60hz ground loop from the device that is calling in. Try using a different outlet, a different device, or if it's a cordless device like a laptop, try it with the power cable unplugged and only use battery to test it. Typically there is nothing we can do being on the receiving end of such noise. 2009/4/1 Stromin Normin > Hi, > > I've been asked to do some testing on Freeswitch by work, we currently use > Asterisk. I'm quite new to telephony so please go easy. > > I have FS setup on a windows box and at the moment I'm testing internal > calls only, when I transfer calls or call extensions everything sounds > great. The problem occurrs when I setup conferencing, people can log in ok > and we can talk, the trouble is as people start to talk a buzzing sound is > heard in the background, once the talking stops the buzzing stops. If the > person goes on mute there is no buzzing. > > Hopefully this is enough info cheers for any help. > > ------------------------------ > " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/b6734373/attachment-0002.html From msc at freeswitch.org Wed Apr 1 14:22:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 14:22:16 -0700 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: References: Message-ID: <87f2f3b90904011422i32e2517dhd5cfc9414468c08@mail.gmail.com> Rupa, Thanks for adding to the project! Well done. -MC 2009/4/1 Rupa Schomaker > Announcing a new module: mod_memcache > > Up until now one had two choices for storing arbitrary key/value pairs. > hash or db. hash is fast, but it is local to the current FreeSWITCH > instance. If you run multiple instances of FreeSWITCH then one could use > db, an ODBC connection and a centralized database server (eg: postgresql). > > The choice was between fast but isolated or slow and distributed. > > memcached (http://www.danga.com/memcached/) is a high-performance, > distributed memory object caching system, generic in nature, but intended > for use in speeding up dynamic web applications by alleviating database > load. Only now you can use it for dynamic phone applications. > > Check out the wiki page at: http://wiki.freeswitch.org/wiki/Mod_memcache > > Try this module out and file bug (jira) reports for problems / enhancement > requests. > > -- > -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/20090401/22f090fe/attachment-0002.html From anthony.minessale at gmail.com Wed Apr 1 14:23:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 16:23:48 -0500 Subject: [Freeswitch-users] Missing CHANNEL_HANGUP event in mod_event_socket In-Reply-To: <1238617166.3750.93.camel@ktulu> References: <1238617166.3750.93.camel@ktulu> Message-ID: <191c3a030904011423r29b5da21m4b5ca9d1fa2099ed@mail.gmail.com> its a race, sometimes the socket connection ends before the channel the linger socket command was added to tell FS to wait for the last channel event before ending the connection just send the command linger On Wed, Apr 1, 2009 at 3:19 PM, Victor Toofic wrote: > Hi all!! > > I'm stuck trying to use mod_event_socket in outbound mode. The problem > that I'm facing is that while in a incoming call, using "myevents" to > monitor for the channel's events.. the event CHANNEL_HANGUP sometimes > arrives and sometimes doesn't. I can't figure it out why. > > The dialplan is: > > > > > > > > > The process that handles the connection does: > > 1. connect > 2. myevents > (received: Reply-Text: +OK Events Enabled) > 3. sendmsg\n call-command: execute\n execute-app-name: answer > (received: Reply-Text: +OK) > > after this it waits for events and/or for the other party to hangup the > call. (The DTMFs are for testing propourses). > > Sometimes the events that the process receives are: > > <<"CHANNEL_PARK">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_ANSWER">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"DTMF">> > <<"DTMF">> > <<"CHANNEL_HANGUP">> > > (then it receives the "text/disconnect-notice" and the socket gets > closed) > > and sometimes are: > > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_ANSWER">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"DTMF">> > <<"DTMF">> > > (then it receives the "text/disconnect-notice" and the socket gets > closed) > > As you can see, even sometimes the first CHANNEL_PARK event doesn't > arrive. I'm very concerned about the missing CHANNEL_HANGUP event. > > In the other hand I was watching the events in a inbound connection to > mod_event_socket with "event text all" and in this case there was no > problem, all the events arrived as expected. > > Why in outbound mode some events get lost?? > I'm missing something?? > > I've tried it in two different machines and the results are the same. > I'm using FreeSWITCH Version 1.0.3 (exported) on linux. > > Thnks!! > > -- > Regards.. > Victor Toofic > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/59e29ec9/attachment-0002.html From anthony.minessale at gmail.com Wed Apr 1 14:24:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Apr 2009 16:24:38 -0500 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: References: Message-ID: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> Thank you, You are brave to contribute something on April 1st =D I saw it go into tree everyone so it's real ;) 2009/4/1 Rupa Schomaker > Announcing a new module: mod_memcache > > Up until now one had two choices for storing arbitrary key/value pairs. > hash or db. hash is fast, but it is local to the current FreeSWITCH > instance. If you run multiple instances of FreeSWITCH then one could use > db, an ODBC connection and a centralized database server (eg: postgresql). > > The choice was between fast but isolated or slow and distributed. > > memcached (http://www.danga.com/memcached/) is a high-performance, > distributed memory object caching system, generic in nature, but intended > for use in speeding up dynamic web applications by alleviating database > load. Only now you can use it for dynamic phone applications. > > Check out the wiki page at: http://wiki.freeswitch.org/wiki/Mod_memcache > > Try this module out and file bug (jira) reports for problems / enhancement > requests. > > -- > -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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/d9f29280/attachment-0002.html From rupa at rupa.com Wed Apr 1 14:31:09 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 1 Apr 2009 16:31:09 -0500 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> Message-ID: 2009/4/1 Anthony Minessale > Thank you, > > You are brave to contribute something on April 1st =D > I saw it go into tree everyone so it's real ;) > haha! I didn't even think of that. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/b00c6fad/attachment-0002.html From jmesquita at gmail.com Wed Apr 1 14:37:47 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 1 Apr 2009 18:37:47 -0300 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> Message-ID: <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> Congrats on the contribution Rupa. And thank you. Mesquita On Apr 1, 2009, at 6:31 PM, Rupa Schomaker wrote: > > > 2009/4/1 Anthony Minessale > Thank you, > > You are brave to contribute something on April 1st =D > I saw it go into tree everyone so it's real ;) > > haha! I didn't even think of that. > > -- > -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/20090401/fcfcf14f/attachment-0002.html From stormin.normin at hotmail.co.uk Wed Apr 1 14:36:15 2009 From: stormin.normin at hotmail.co.uk (Stromin Normin) Date: Wed, 1 Apr 2009 22:36:15 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: Cheers for the replies. I'm not sure if I'm replying properly but here goes. I'm using Polycom 650 phones. I'm not really sure what a 60hz ground loop is so will need clarification, sorry I'm new to this. The phones are all on the same LAN and the conferencing is done on internal calls. Cheers From: stormin.normin at hotmail.co.uk To: freeswitch-users at lists.freeswitch.org Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now _________________________________________________________________ View your Twitter and Flickr updates from one place ? Learn more! http://clk.atdmt.com/UKM/go/137984870/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/88ba5c77/attachment-0002.html From gmaruzz at celliax.org Wed Apr 1 15:10:35 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 2 Apr 2009 00:10:35 +0200 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: <7b197bef0904011510lb8adaadka391debfa2ec2a91@mail.gmail.com> To make a long story short, a ground loop is when an electric circuit is made between different audio device that are connected to the same electric power grid with badly grounded connections. This is an electrical problem generating noise, nothing to do with software. To test if this is the origin of your problem, try to use the devices unplugged from the electrical grid and check if the noise still there Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 2009/4/1 Stromin Normin : > Cheers for the replies.? I'm not sure if I'm replying properly but here > goes. > > I'm using Polycom 650 phones. > > I'm not really sure what a 60hz ground loop is so will need clarification, > sorry I'm new to this.? The phones are all on the same LAN and the > conferencing is done on internal calls. > > Cheers > > ________________________________ > From: stormin.normin at hotmail.co.uk > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 1 Apr 2009 22:09:03 +0100 > Subject: [Freeswitch-users] Buzzing when people speak in conference > > Hi, > > I've been asked to do some testing on Freeswitch by work, we currently use > Asterisk.? I'm quite new to telephony so please go easy. > > I have FS setup on a windows box and at the moment I'm testing internal > calls only, when I transfer calls or call extensions everything sounds > great.? The problem occurrs when I setup conferencing, people can log in ok > and we can talk, the trouble is as people start to talk a buzzing sound is > heard in the background, once the talking stops the buzzing stops.? If the > person goes on mute there is no buzzing. > > Hopefully this is enough info cheers for any help. > > ________________________________ > " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now > ________________________________ > Surfing the web just got more rewarding. Download the New Internet Explorer > 8 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Wed Apr 1 15:29:32 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 1 Apr 2009 17:29:32 -0500 Subject: [Freeswitch-users] mod_shout delay in trunk Message-ID: I've setup a conference bridge that has perpetual-sound set to a icecast stream. When the first person connects, there is an ~7s delay before any audio is heard. This is similar to a problem reported by Dan here and concluded with Tony adding the channel var "enable_file_write_buffering". The list discussion ended here: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011104.html I set this var in my dialplan: prior to joining the conference. The first person in still sees a 7s delay on audio the first time in. Like dan, I have icecast setup with burst_on_connect set to 1 but my burst_size is the default 64k so quite a bit of data. Has anyone been able to get an on-demand shoutcast stream from an icecast server to start immediately (or at least within a second)? Thanks. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/0faf0533/attachment-0002.html From thorhs at basis.is Wed Apr 1 15:34:30 2009 From: thorhs at basis.is (Thorhallur Sverrisson) Date: Wed, 01 Apr 2009 22:34:30 +0000 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: <49D3EBF6.5050808@basis.is> The Polycom 650 is an IP phone, so the ground loop should not apply. Ground loops occur only in analog systems. As to what the buzzing is, I'm not sure. I have performed tests using Polycom 650s with out any sound artifacts. In fact the 650 audio has been flawless in my testing. Sorry I don't have a solution, just wanted to steer you away from a ground-loop debugging session. Thorhallur Stromin Normin wrote: > Cheers for the replies. I'm not sure if I'm replying properly but here > goes. > > I'm using Polycom 650 phones. > > I'm not really sure what a 60hz ground loop is so will need > clarification, sorry I'm new to this. The phones are all on the same > LAN and the conferencing is done on internal calls. > > Cheers > From msc at freeswitch.org Wed Apr 1 15:44:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Apr 2009 15:44:07 -0700 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: <49D3EBF6.5050808@basis.is> References: <49D3EBF6.5050808@basis.is> Message-ID: <87f2f3b90904011544t6d58c510ue707c98ac492bd2a@mail.gmail.com> That being the case, maybe a pcap of the audio might yield some clues? On Wed, Apr 1, 2009 at 3:34 PM, Thorhallur Sverrisson wrote: > The Polycom 650 is an IP phone, so the ground loop should not apply. > Ground loops occur only in analog systems. > > As to what the buzzing is, I'm not sure. I have performed tests using > Polycom 650s with out any sound artifacts. In fact the 650 audio has > been flawless in my testing. > > Sorry I don't have a solution, just wanted to steer you away from a > ground-loop debugging session. > > Thorhallur > > > Stromin Normin wrote: > > Cheers for the replies. I'm not sure if I'm replying properly but here > > goes. > > > > I'm using Polycom 650 phones. > > > > I'm not really sure what a 60hz ground loop is so will need > > clarification, sorry I'm new to this. The phones are all on the same > > LAN and the conferencing is done on internal calls. > > > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/78c9534f/attachment-0002.html From hads at nice.net.nz Wed Apr 1 15:53:19 2009 From: hads at nice.net.nz (Hadley Rich) Date: Thu, 2 Apr 2009 11:53:19 +1300 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: <49D3EBF6.5050808@basis.is> References: <49D3EBF6.5050808@basis.is> Message-ID: <200904021153.19827.hads@nice.net.nz> On Thu, 02 Apr 2009 11:34:30 Thorhallur Sverrisson wrote: > The Polycom 650 is an IP phone, so the ground loop should not apply. > Ground loops occur only in analog systems. There is always an analog part to the system thus the potential for ground loops. It's common with snom phones when using a headset but I've not seen an issue with Polycom yet. hads -- http://nicegear.co.nz VoIP, DVB and other Linux compatible hardware. From elhodred at gmail.com Wed Apr 1 15:36:28 2009 From: elhodred at gmail.com (Alfonso Pinto) Date: Thu, 2 Apr 2009 00:36:28 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls Message-ID: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> Hi guys, I've using asterisk as PSTN gateway. When a call arrives from PSTN, I send the call to freeswitch and this route the call to a SIP gateway. When caller cancels the call (hangups before callee answers), I get this on asterisk CLI: chan_sip.c:13056 handle_response: Remote host can't match request CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 This is the sip call flow: u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 INVITE sip:666666666 at 1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "999999999" ;tag=as26208773. To: . Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Wed, 01 Apr 2009 21:03:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 29347 29347 IN IP4 2.2.2.2. s=session. c=IN IP4 2.2.2.2. t=0 0. m=audio 13846 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. From: "999999999" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Content-Length: 0. . U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. From: "999999999" ;tag=as26208773. To: ;tag=ceKFmNU84B90c. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Proxy-Authenticate: Digest realm="1.1.1.1", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, qop="auth". Content-Length: 0. . U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:666666666 at 1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "999999999" ;tag=as26208773. To: ;tag=ceKFmNU84B90c. Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 INVITE sip:666666666 at 1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. From: "999999999" ;tag=as26208773. To: . Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", algorithm=MD5, uri="sip:666666666 at 1.1.1.1", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, cnonce="47efcad4", nc=00000001. Date: Wed, 01 Apr 2009 21:03:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 29347 29348 IN IP4 2.2.2.2. s=session. c=IN IP4 2.2.2.2. t=0 0. m=audio 13846 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: "999999999" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Content-Length: 0. . U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 INVITE sip:666666666 at 3.3.3.3 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. Max-Forwards: 69. From: "999999999" ;tag=e050QBXFZXN6K. To: . Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 387. Remote-Party-ID: "999999999" ;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. s=FreeSWITCH. c=IN IP4 1.1.1.1. t=0 0. m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:3 GSM/8000. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:46:57 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 113193247 INVITE. Allow-Events: telephone-event. Content-Length: 0. . U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:46:57 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 113193247 INVITE. Allow-Events: telephone-event. Contact: . Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 300. . v=0. o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. s=SIP Call. c=IN IP4 3.3.3.3. t=0 0. m=audio 19398 RTP/AVP 18 13 101. c=IN IP4 3.3.3.3. a=rtpmap:18 G729/8000. a=rtpmap:13 CN/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:40. U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 292. . v=0. o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. s=FreeSWITCH. c=IN IP4 1.1.1.1. t=0 0. m=audio 20620 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. From: "999999999" ;tag=as26208773. To: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 CANCEL. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 481 Call/Transaction Does Not Exist. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 CANCEL. Content-Length: 0. . U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:46:57 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 113193247 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: . Content-Type: application/sdp. Content-Length: 300. . v=0. o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. s=SIP Call. c=IN IP4 3.3.3.3. t=0 0. m=audio 19398 RTP/AVP 18 13 101. c=IN IP4 3.3.3.3. a=rtpmap:18 G729/8000. a=rtpmap:13 CN/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:40. U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. Max-Forwards: 70. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 ACK. Contact: . Content-Length: 0. . U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 292. . v=0. o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. s=FreeSWITCH. c=IN IP4 1.1.1.1. t=0 0. m=audio 20620 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Contact: . Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 103 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 104 BYE. User-Agent: Asterisk PBX. Max-Forwards: 70. Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, cnonce="6bc43301", nc=00000002. Content-Length: 0. . U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. From: "999999999" ;tag=as26208773. To: ;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. CSeq: 104 BYE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Content-Length: 0. . U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. Max-Forwards: 70. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193248 BYE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. . U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. From: "999999999" ;tag=e050QBXFZXN6K. To: ;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:47:32 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. Content-Length: 0. CSeq: 113193248 BYE. . Please, can somebody tell me what is happening?. Thanks in advance. Regards. From chris at cloudtel.com Wed Apr 1 16:29:36 2009 From: chris at cloudtel.com (Chris Burns) Date: Wed, 1 Apr 2009 16:29:36 -0700 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: <7b197bef0904011510lb8adaadka391debfa2ec2a91@mail.gmail.com> References: <7b197bef0904011510lb8adaadka391debfa2ec2a91@mail.gmail.com> Message-ID: <200904011629.36433.chris@cloudtel.com> Try turning off comfort noise completely in the conference profile? My 650s sound great in conference w/ PCMU and G722 On April 1, 2009 03:10:35 pm Giovanni Maruzzelli wrote: > To make a long story short, a ground loop is when an electric circuit > is made between different audio device that are connected to the same > electric power grid with badly grounded connections. > > This is an electrical problem generating noise, nothing to do with > software. > > To test if this is the origin of your problem, try to use the devices > unplugged from the electrical grid and check if the noise still there > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > 2009/4/1 Stromin Normin : > > Cheers for the replies.? I'm not sure if I'm replying properly but here > > goes. > > > > I'm using Polycom 650 phones. > > > > I'm not really sure what a 60hz ground loop is so will need > > clarification, sorry I'm new to this.? The phones are all on the same LAN > > and the conferencing is done on internal calls. > > > > Cheers > > > > ________________________________ > > From: stormin.normin at hotmail.co.uk > > To: freeswitch-users at lists.freeswitch.org > > Date: Wed, 1 Apr 2009 22:09:03 +0100 > > Subject: [Freeswitch-users] Buzzing when people speak in conference > > > > Hi, > > > > I've been asked to do some testing on Freeswitch by work, we currently > > use Asterisk.? I'm quite new to telephony so please go easy. > > > > I have FS setup on a windows box and at the moment I'm testing internal > > calls only, when I transfer calls or call extensions everything sounds > > great.? The problem occurrs when I setup conferencing, people can log in > > ok and we can talk, the trouble is as people start to talk a buzzing > > sound is heard in the background, once the talking stops the buzzing > > stops.? If the person goes on mute there is no buzzing. > > > > Hopefully this is enough info cheers for any help. > > > > ________________________________ > > " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now > > ________________________________ > > Surfing the web just got more rewarding. Download the New Internet > > Explorer 8 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 16:34:51 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 18:34:51 -0500 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> Message-ID: I'm pretty sure this is a bug in Asterisk something to do with dialog matching... I think if you search the archives you'll see about it. /b On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666 at 1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666 at 3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" 999999999 at 3.3.3.3>;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: . > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: . > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: . > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > 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 Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/3d470bd8/attachment-0002.html From elhodred at gmail.com Wed Apr 1 16:41:57 2009 From: elhodred at gmail.com (Alfonso Pinto) Date: Thu, 2 Apr 2009 01:41:57 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> Message-ID: <8b3b7acc0904011641v2c7e3768ne72e06623ed064f2@mail.gmail.com> I've searched in google about it and only found a message about the same, Anthony asked for more information and nobody answer. I've tried with an IP phone (aastra 57i) and the same happens. Thank you 2009/4/2 Brian West : > I'm pretty sure this is a bug in Asterisk something to do with dialog > matching... I think if you search the archives you'll see about it. > /b > On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the ?call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666 at 1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666 at 3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" ;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" ;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: . > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: . > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: . > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > 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 > > Brian West > brian at freeswitch.org > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 16:46:35 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 18:46:35 -0500 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> Message-ID: <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> Follow this thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html /b On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666 at 1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666 at 3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" 999999999 at 3.3.3.3>;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: . > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: . > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: . > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > 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 Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/a9fd1495/attachment-0002.html From elhodred at gmail.com Wed Apr 1 17:09:42 2009 From: elhodred at gmail.com (Alfonso Pinto) Date: Thu, 2 Apr 2009 02:09:42 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> Message-ID: <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> One question more, maybe a stupid one: How can I search the archives? I didn't find nothing in lists.freeswitch.org. Regards 2009/4/2 Brian West : > Follow this > thread?http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html > /b > On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the ?call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666 at 1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666 at 3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" ;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" ;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: . > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: . > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: . > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > 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 > > Brian West > brian at freeswitch.org > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 17:19:06 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 19:19:06 -0500 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> Message-ID: If you go to google and input "site:lists.freeswitch.org blah" /b On Apr 1, 2009, at 7:09 PM, Alfonso Pinto wrote: > One question more, maybe a stupid one: How can I search the archives? > I didn't find nothing in lists.freeswitch.org. > > Regards Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/7e0d16cc/attachment-0002.html From jason at jasonjgw.net Wed Apr 1 17:35:33 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 2 Apr 2009 11:35:33 +1100 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> Message-ID: <20090402003533.GA9849@jdc.jasonjgw.net> Alfonso Pinto wrote: > One question more, maybe a stupid one: How can I search the archives? http://www.gmane.org/ The searching tool they use, Xapian, tends to give good relevance ranking, at least in my experience. From sicfslist at gmail.com Wed Apr 1 17:56:32 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 1 Apr 2009 19:56:32 -0500 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: <200904011629.36433.chris@cloudtel.com> References: <7b197bef0904011510lb8adaadka391debfa2ec2a91@mail.gmail.com> <200904011629.36433.chris@cloudtel.com> Message-ID: <35b355e90904011756i3b3192fcm582b7e966e2397fb@mail.gmail.com> I have in a previous life seen this quite a bit with the PolyCom phones ... people tend to put their phone on the speaker on conference calls and I have seen this type of interference caused by a computer speaker and even a motorola cell phone. So I would first force everyone to use the handset 1st ... if that solves it then track down the guilty speaker or cell phone. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/9078884d/attachment-0002.html From sicfslist at gmail.com Wed Apr 1 17:59:40 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 1 Apr 2009 19:59:40 -0500 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> Message-ID: <35b355e90904011759s43c91d4cg4d7a95b66d8027bd@mail.gmail.com> Rupa, This is a big contribution! Thanks! Can't wait to play with this. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/85c572d1/attachment-0002.html From toofics at gmail.com Wed Apr 1 17:44:13 2009 From: toofics at gmail.com (Victor Toofic) Date: Wed, 01 Apr 2009 18:44:13 -0600 Subject: [Freeswitch-users] Missing CHANNEL_HANGUP event in mod_event_socket In-Reply-To: <191c3a030904011423r29b5da21m4b5ca9d1fa2099ed@mail.gmail.com> References: <1238617166.3750.93.camel@ktulu> <191c3a030904011423r29b5da21m4b5ca9d1fa2099ed@mail.gmail.com> Message-ID: <1238633053.3750.99.camel@ktulu> thnks a lot!! I was getting scared.. lol Freeswitch rules!! On Wed, 2009-04-01 at 16:23 -0500, Anthony Minessale wrote: > its a race, > > sometimes the socket connection ends before the channel > > the linger socket command was added to tell FS to wait for the last > channel event before > ending the connection > > just send the command > > linger > > > > On Wed, Apr 1, 2009 at 3:19 PM, Victor Toofic > wrote: > Hi all!! > > I'm stuck trying to use mod_event_socket in outbound mode. The > problem > that I'm facing is that while in a incoming call, using > "myevents" to > monitor for the channel's events.. the event CHANNEL_HANGUP > sometimes > arrives and sometimes doesn't. I can't figure it out why. > > The dialplan is: > > > > > > > > > The process that handles the connection does: > > 1. connect > 2. myevents > (received: Reply-Text: +OK Events Enabled) > 3. sendmsg\n call-command: execute\n execute-app-name: answer > (received: Reply-Text: +OK) > > after this it waits for events and/or for the other party to > hangup the > call. (The DTMFs are for testing propourses). > > Sometimes the events that the process receives are: > > <<"CHANNEL_PARK">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_ANSWER">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"DTMF">> > <<"DTMF">> > <<"CHANNEL_HANGUP">> > > (then it receives the "text/disconnect-notice" and the socket > gets > closed) > > and sometimes are: > > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"CHANNEL_EXECUTE">> > <<"CHANNEL_ANSWER">> > <<"CHANNEL_EXECUTE_COMPLETE">> > <<"DTMF">> > <<"DTMF">> > > (then it receives the "text/disconnect-notice" and the socket > gets > closed) > > As you can see, even sometimes the first CHANNEL_PARK event > doesn't > arrive. I'm very concerned about the missing CHANNEL_HANGUP > event. > > In the other hand I was watching the events in a inbound > connection to > mod_event_socket with "event text all" and in this case there > was no > problem, all the events arrived as expected. > > Why in outbound mode some events get lost?? > I'm missing something?? > > I've tried it in two different machines and the results are > the same. > I'm using FreeSWITCH Version 1.0.3 (exported) on linux. > > Thnks!! > > -- > Regards.. > Victor Toofic > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 1 18:06:19 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Apr 2009 20:06:19 -0500 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: <35b355e90904011759s43c91d4cg4d7a95b66d8027bd@mail.gmail.com> References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> <35b355e90904011759s43c91d4cg4d7a95b66d8027bd@mail.gmail.com> Message-ID: <7C5053E3-1F30-4D6C-B786-004931E813A1@freeswitch.org> At the very least you didn't say "I can't wait to play with it!" :P On Apr 1, 2009, at 7:59 PM, Shelby Ramsey wrote: > Rupa, > > This is a big contribution! Thanks! Can't wait to play with this. > > SDR Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/9cf25d0e/attachment-0002.html From kristian.kielhofner at gmail.com Wed Apr 1 22:17:51 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 2 Apr 2009 01:17:51 -0400 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> References: <5800526b0904011140y45412ff3wb120c85b7e893f46@mail.gmail.com> Message-ID: <2d9149cd0904012217s692c4666tccb25b0db70b498b@mail.gmail.com> Carlos, I'm glad to see you've made some progress on your project. Keep us updated! 2009/4/1 Carlos Talbot : > > Is there an interest in running FreeSWITCH on OpenWRT? I recently managed to > compile and run a version for a MIPs based router, the Planex MZK-W04NU. > This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support > (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks > online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP > space. > > I was planning to setup a wiki page on compiling and configuring. > > regards, > > Carlos > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Wed Apr 1 23:09:56 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 2 Apr 2009 02:09:56 -0400 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <8b3b7acc0904011641v2c7e3768ne72e06623ed064f2@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <8b3b7acc0904011641v2c7e3768ne72e06623ed064f2@mail.gmail.com> Message-ID: <2d9149cd0904012309h5428ece5qce9a287b0972dd81@mail.gmail.com> I probably shouldn't be doing this for you, but... http://bugs.digium.com/view.php?id=14431 ;) On Wed, Apr 1, 2009 at 7:41 PM, Alfonso Pinto wrote: > I've searched in google about it and only found a message about the > same, Anthony asked for more information and nobody answer. > > I've tried with an IP phone (aastra 57i) and the same happens. > > Thank you > > 2009/4/2 Brian West : >> I'm pretty sure this is a bug in Asterisk something to do with dialog >> matching... I think if you search the archives you'll see about it. >> /b >> On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: >> >> Hi guys, >> >> I've using asterisk as PSTN gateway. When a call arrives from PSTN, I >> send the call to freeswitch and this route the call to a SIP gateway. >> >> When caller cancels the ?call (hangups before callee answers), I get >> this on asterisk CLI: >> >> chan_sip.c:13056 handle_response: Remote host can't match request >> CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. >> >> I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 >> >> This is the sip call flow: >> >> u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 >> INVITE sip:666666666 at 1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. >> From: "999999999" ;tag=as26208773. >> To: . >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 102 INVITE. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Date: Wed, 01 Apr 2009 21:03:12 GMT. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. >> Supported: replaces. >> Content-Type: application/sdp. >> Content-Length: 265. >> . >> v=0. >> o=root 29347 29347 IN IP4 2.2.2.2. >> s=session. >> c=IN IP4 2.2.2.2. >> t=0 0. >> m=audio 13846 RTP/AVP 18 101. >> a=rtpmap:18 G729/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=silenceSupp:off - - - -. >> a=ptime:20. >> a=sendrecv. >> >> >> U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 100 Trying. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 102 INVITE. >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 407 Proxy Authentication Required. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: ;tag=ceKFmNU84B90c. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 102 INVITE. >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Accept: application/sdp. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer. >> Proxy-Authenticate: Digest realm="1.1.1.1", >> nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, >> qop="auth". >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 >> ACK sip:666666666 at 1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. >> From: "999999999" ;tag=as26208773. >> To: ;tag=ceKFmNU84B90c. >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 102 ACK. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 >> INVITE sip:666666666 at 1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. >> From: "999999999" ;tag=as26208773. >> To: . >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 INVITE. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", >> algorithm=MD5, uri="sip:666666666 at 1.1.1.1", >> nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", >> response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, >> cnonce="47efcad4", nc=00000001. >> Date: Wed, 01 Apr 2009 21:03:12 GMT. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. >> Supported: replaces. >> Content-Type: application/sdp. >> Content-Length: 265. >> . >> v=0. >> o=root 29347 29348 IN IP4 2.2.2.2. >> s=session. >> c=IN IP4 2.2.2.2. >> t=0 0. >> m=audio 13846 RTP/AVP 18 101. >> a=rtpmap:18 G729/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=silenceSupp:off - - - -. >> a=ptime:20. >> a=sendrecv. >> >> >> U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 100 Trying. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 INVITE. >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 >> INVITE sip:666666666 at 3.3.3.3 SIP/2.0. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. >> Max-Forwards: 69. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: . >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> CSeq: 113193247 INVITE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer. >> Content-Type: application/sdp. >> Content-Disposition: session. >> Content-Length: 387. >> Remote-Party-ID: "999999999" ;screen=yes;privacy=off. >> . >> v=0. >> o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. >> s=FreeSWITCH. >> c=IN IP4 1.1.1.1. >> t=0 0. >> m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. >> a=rtpmap:18 G729/8000. >> a=rtpmap:4 G723/8000. >> a=rtpmap:3 GSM/8000. >> a=rtpmap:9 G722/8000. >> a=rtpmap:0 PCMU/8000. >> a=rtpmap:8 PCMA/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=rtpmap:13 CN/8000. >> a=ptime:20. >> >> >> U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 >> SIP/2.0 100 Trying. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Date: Fri, 05 Jan 2001 07:46:57 GMT. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> Server: Cisco-SIPGateway/IOS-12.x. >> CSeq: 113193247 INVITE. >> Allow-Events: telephone-event. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 >> SIP/2.0 183 Session Progress. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Date: Fri, 05 Jan 2001 07:46:57 GMT. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> Server: Cisco-SIPGateway/IOS-12.x. >> CSeq: 113193247 INVITE. >> Allow-Events: telephone-event. >> Contact: . >> Content-Disposition: session;handling=required. >> Content-Type: application/sdp. >> Content-Length: 300. >> . >> v=0. >> o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. >> s=SIP Call. >> c=IN IP4 3.3.3.3. >> t=0 0. >> m=audio 19398 RTP/AVP 18 13 101. >> c=IN IP4 3.3.3.3. >> a=rtpmap:18 G729/8000. >> a=rtpmap:13 CN/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:40. >> >> >> U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 183 Session Progress. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 INVITE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Accept: application/sdp. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer. >> Content-Type: application/sdp. >> Content-Disposition: session. >> Content-Length: 292. >> . >> v=0. >> o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. >> s=FreeSWITCH. >> c=IN IP4 1.1.1.1. >> t=0 0. >> m=audio 20620 RTP/AVP 18 101. >> a=rtpmap:18 G729/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=silenceSupp:off - - - -. >> a=ptime:20. >> >> >> U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 >> CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. >> From: "999999999" ;tag=as26208773. >> To: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 CANCEL. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 481 Call/Transaction Does Not Exist. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 CANCEL. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Date: Fri, 05 Jan 2001 07:46:57 GMT. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> Server: Cisco-SIPGateway/IOS-12.x. >> CSeq: 113193247 INVITE. >> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, >> SUBSCRIBE, NOTIFY, INFO. >> Allow-Events: telephone-event. >> Contact: . >> Content-Type: application/sdp. >> Content-Length: 300. >> . >> v=0. >> o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. >> s=SIP Call. >> c=IN IP4 3.3.3.3. >> t=0 0. >> m=audio 19398 RTP/AVP 18 13 101. >> c=IN IP4 3.3.3.3. >> a=rtpmap:18 G729/8000. >> a=rtpmap:13 CN/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:40. >> >> >> U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 >> ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. >> Max-Forwards: 70. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> CSeq: 113193247 ACK. >> Contact: . >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 INVITE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer. >> Content-Type: application/sdp. >> Content-Disposition: session. >> Content-Length: 292. >> . >> v=0. >> o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. >> s=FreeSWITCH. >> c=IN IP4 1.1.1.1. >> t=0 0. >> m=audio 20620 RTP/AVP 18 101. >> a=rtpmap:18 G729/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=silenceSupp:off - - - -. >> a=ptime:20. >> >> >> U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 >> ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Contact: . >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 103 ACK. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 >> BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 104 BYE. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", >> algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", >> nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", >> response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, >> cnonce="6bc43301", nc=00000002. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. >> From: "999999999" ;tag=as26208773. >> To: ;tag=DQc8Ngcc2mZKr. >> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. >> CSeq: 104 BYE. >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 >> BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. >> Max-Forwards: 70. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> CSeq: 113193248 BYE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Reason: Q.850;cause=16;text="NORMAL_CLEARING". >> Content-Length: 0. >> . >> >> >> U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. >> From: "999999999" ;tag=e050QBXFZXN6K. >> To: ;tag=731C8E54-1862. >> Date: Fri, 05 Jan 2001 07:47:32 GMT. >> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. >> Server: Cisco-SIPGateway/IOS-12.x. >> Content-Length: 0. >> CSeq: 113193248 BYE. >> . >> >> Please, can somebody tell me what is happening?. >> >> Thanks in advance. >> >> 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 >> >> Brian West >> brian at freeswitch.org >> -- Meet us a ClueCon! ?http://www.cluecon.com >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From ashley.ohq at gmail.com Thu Apr 2 00:08:44 2009 From: ashley.ohq at gmail.com (Ashley van Gerven) Date: Thu, 2 Apr 2009 18:08:44 +1100 Subject: [Freeswitch-users] FS failover redundancy & load balancing Message-ID: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> Hi, I can't find much info on setting up a redundant or heavy load FreeSwitch implementation. Are there any links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? I imagine the entry level solution is to have two FS boxes configured identitcally, with redundant SBC software (recommendations?) in front, passing the calls to the primary FS box, or the backup FS box if the primary is not responding. Is that the easiest solution? What about a situation of having a level of concurrent calls beyond what one FS box can handle? I realise that would be a very large number of concurrent calls, but we would need a good plan on how to scale the systems. Are there recommendations for load balancing solutions? Either soft or hardware? My guess would be having 3 + 1 spare FS servers would work, where calls are distributed accross 3 FS boxes by a load balancer with one spare in event of failure. Also how would a FS box at max capacity behave? Does FS monitor available resources and reject the excess calls that it can't handle? Or would the load balancer have to be configured with the maximum number of calls per box? Would love to hear some experiences of deploying FS with failover & high load. Thanks Ash -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/807cb762/attachment-0002.html From gmaruzz at celliax.org Thu Apr 2 01:35:21 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 2 Apr 2009 10:35:21 +0200 Subject: [Freeswitch-users] Skype interaction commands on skypiax Message-ID: <7b197bef0904020135j6b56662dy5a0dd2862ac4f35d@mail.gmail.com> Hi all, background: mod_skypiax is Skype compatible endpoint that allows incoming and outbound calls to/from the Skype network and SkypeOut service. It's seen by FS like other endpoints, so you can originate from sofia, bridge to skypiax, and connect the call to a landline number via SkypeOut service, for eg. skypiax endpoint use a Skype client to interact with the Skype network (see the wiki page for more details http://wiki.freeswitch.org/wiki/Skypiax). The news are: now you can send commands to the skype client related to a skyiax interface, both from the FS command line and programmatically (socket/API/esl/whatever) http://wiki.freeswitch.org/wiki/Skypiax#API_and_CLI_Commands This allow you to use directly the entire power of the Skype API ( https://developer.skype.com/Docs/ApiDoc ), for eg to send chat messages, interact with the buddy list, etc etc. Typing "console loglevel 9" at the FS command line allows you to see the Skype API answers from the Skype client instance. So, in short: you bring loglevel to 9 (so you can see the Skype API messages going back and forth), you use "sk" or "skypiax" to send Skype API commands to the Skype client instance. This way you can prototype extensions to the current mod_skypiax, that can then be implemented in C directly into the mod_skypiax source code. Please, let me know of extensions you would like to be integrated into the mod_skypiax code ;-). Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 From tristan at telemaque.fr Thu Apr 2 02:01:09 2009 From: tristan at telemaque.fr (Tristan) Date: Thu, 02 Apr 2009 11:01:09 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> Message-ID: <49D47ED5.5020409@telemaque.fr> Hi Ashley, One easy solution is to use a SIP proxy (opensips/kamailio/...) in front of FS boxes to load balance the charge between boxes. FS already has mechanisms to limit number of calls per boxes ( in switch.conf.xml: max-sessions and sessions-per-second ), that you can couple to load_balancing modules of the sip proxies. Of course you'll have to test to know how many session one box can handle, has it depends a lot on your usage of FS. Don' hesitate to join us on IRC if you want to discuss it ;) Regards, Gled Ashley van Gerven a ?crit : > Hi, > > I can't find much info on setting up a redundant or heavy load > FreeSwitch implementation. Are there any > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? > > I imagine the entry level solution is to have two FS boxes configured > identitcally, with > redundant SBC software (recommendations?) in front, passing the calls > to the primary FS box, > or the backup FS box if the primary is not responding. Is that the > easiest solution? > > What about a situation of having a level of concurrent calls beyond > what one FS box can handle? I realise > that would be a very large number of concurrent calls, but we would > need a good plan on how to scale the > systems. > > Are there recommendations for load balancing solutions? Either soft or > hardware? > > My guess would be having 3 + 1 spare FS servers would work, where > calls are distributed accross 3 FS boxes > by a load balancer with one spare in event of failure. > > Also how would a FS box at max capacity behave? Does FS monitor > available resources and reject the > excess calls that it can't handle? Or would the load balancer have to > be configured with the maximum number > of calls per box? > > Would love to hear some experiences of deploying FS with failover & > high load. > > > Thanks > Ash > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/c2aa1062/attachment-0002.html From sridhart at alcatel-lucent.com Thu Apr 2 01:58:30 2009 From: sridhart at alcatel-lucent.com (Rajagopal, Sridhar (Sridhar)) Date: Thu, 2 Apr 2009 14:28:30 +0530 Subject: [Freeswitch-users] Dialplan for OPTIONS packet Message-ID: <9389DD3DDD6B9144B147CE564C6599B902D22BA2C9@INBANSXCHMBSA3.in.alcatel-lucent.com> Hi all, Whenever freeswitch recieves INVITE SIP packet, It forwards the packet based on the dial plan. I want to use the same dial plan to forward incoming OPTIONS packet. Please let me know If I need to write my own code for that or is there any such option in our code base. Regards, Sridhar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/d297f341/attachment-0002.html From solko at gcdf.pl Thu Apr 2 02:13:48 2009 From: solko at gcdf.pl (Szymon Olko) Date: Thu, 02 Apr 2009 11:13:48 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> Message-ID: <49D481CC.70102@gcdf.pl> Ashley van Gerven pisze: > Hi, > > I can't find much info on setting up a redundant or heavy load > FreeSwitch implementation. Are there any > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? > > I imagine the entry level solution is to have two FS boxes configured > identitcally, with > redundant SBC software (recommendations?) in front, passing the calls to > the primary FS box, > or the backup FS box if the primary is not responding. Is that the > easiest solution? > > What about a situation of having a level of concurrent calls beyond what > one FS box can handle? I realise > that would be a very large number of concurrent calls, but we would need > a good plan on how to scale the > systems. > > Are there recommendations for load balancing solutions? Either soft or > hardware? > > My guess would be having 3 + 1 spare FS servers would work, where calls > are distributed accross 3 FS boxes > by a load balancer with one spare in event of failure. > > Also how would a FS box at max capacity behave? Does FS monitor > available resources and reject the > excess calls that it can't handle? Or would the load balancer have to be > configured with the maximum number > of calls per box? I did not think yet about HA nor LB. I tested how FS handles high load. All my calls are placed in mod_conference. When cpu usage gets it's limits then new calls can be placed but sound quality is getting worst with every next call. When calls are hanged up then sound gets better. I did not test it to see what happens when more and more calls are created. FS has very low memory consumption and I think that CPU is the limit. I did not notice any monitoring of CPU usage by FS, but my installation is limited to only few modules, so maybe I'm missing something. > > Would love to hear some experiences of deploying FS with failover & high My failover is currently made by shell script which every 10 seconds check for working FS and restarts it if it does not work. I use svn trunk so crash happens once a while, but they are successfully fixed by developers. Once there was a problem that conference module was stuck and did not respond to my commands. I made script with netcat which checks once a while for response and restarts if there is none. > load. > > > Thanks > Ash > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 2 04:55:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 06:55:02 -0500 Subject: [Freeswitch-users] Dialplan for OPTIONS packet In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B902D22BA2C9@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B902D22BA2C9@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <43B08789-DAE7-461A-BA73-3C73B9EAB7DC@freeswitch.org> Can you describe the reasoning behind needing to route option packets via the dialplan? /b On Apr 2, 2009, at 3:58 AM, Rajagopal, Sridhar (Sridhar) wrote: > Hi all, > > Whenever freeswitch recieves INVITE SIP packet, It forwards the > packet based on the dial plan. I want to use the same dial plan to > forward incoming OPTIONS packet. Please let me know If I need to > write my own code for that or is there any such option in our code > base. > > Regards, > Sridhar > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/d2e96f28/attachment-0002.html From bmsword at gmail.com Thu Apr 2 00:29:14 2009 From: bmsword at gmail.com (bmsword) Date: Thu, 2 Apr 2009 15:29:14 +0800 Subject: [Freeswitch-users] about freeswitch conference References: <200904021524116567464@gmail.com> Message-ID: <200904021529137966712@gmail.com> hi,all I want to use another softswitch conference that has been deployed in freeswitch,How should I do? andy 2009-04-02 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/191beb1a/attachment-0002.html From stormin.normin at hotmail.co.uk Thu Apr 2 02:20:25 2009 From: stormin.normin at hotmail.co.uk (Stromin Normin) Date: Thu, 2 Apr 2009 10:20:25 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: Thanks for taking the time to help me. Giovanni, I assume you turn comfort noise off by setting it to 0 which I've now done. How can I tell which codecs I'm using in conference and how would I change them. The sound is ok on everything else. Thanks again From: stormin.normin at hotmail.co.uk To: freeswitch-users at lists.freeswitch.org Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now _________________________________________________________________ View your Twitter and Flickr updates from one place ? Learn more! http://clk.atdmt.com/UKM/go/137984870/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/c3db81ee/attachment-0002.html From bmsword at gmail.com Thu Apr 2 02:26:29 2009 From: bmsword at gmail.com (bmsword) Date: Thu, 2 Apr 2009 17:26:29 +0800 Subject: [Freeswitch-users] about freeswitch conference References: <200904021524116567464@gmail.com> Message-ID: <200904021726283757151@gmail.com> hi,all I want to use another softswitch conference that has been deployed in freeswitch,How should I do? thanks! andy 2009-04-02 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/5a03ddc8/attachment-0002.html From yivzhenko at mksat.net Thu Apr 2 03:22:05 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko (WP)) Date: Thu, 2 Apr 2009 13:22:05 +0300 Subject: [Freeswitch-users] Use rates from lcr in nibblebill module Message-ID: <200904021322.05690.yivzhenko@mksat.net> Hi, I want to use module lcr to find a best route and his rate , then make a call and bill on that rate with nibblebill module. lcr return variable "lcr_auto_route" that contains "[lcr_rate=xxx]" variable for new channel. To use nibblebill i need to set "nibble_rate" = "lcr_rate". What is best method to do that? Is there a way to do that with standard tools, without use external scripts? Thanks, Yuriy From brian at freeswitch.org Thu Apr 2 04:58:58 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 06:58:58 -0500 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <49D481CC.70102@gcdf.pl> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> Message-ID: <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> On Apr 2, 2009, at 4:13 AM, Szymon Olko wrote: > I did not think yet about HA nor LB. > > I tested how FS handles high load. All my calls are placed in > mod_conference. When cpu usage gets it's limits then new calls can > be placed but sound quality is getting worst with every next call. > When calls are hanged up then sound gets better. I did not test > it to see what happens when more and more calls are created. > FS has very low memory consumption and I think that CPU is the > limit. I did not notice any monitoring of CPU usage by FS, but my > installation is limited to only few modules, so maybe I'm missing > something. Load testing against the conference module is about the worst thing you can do... tossing 100+ people in the same conference isn't going to scale well for load testing because its not something you usually do in a real world scenario. Usually you'll have most of the participants muted. I highly recommend you try doing something like a bridge or a file playback from a ram disk. >> >> Would love to hear some experiences of deploying FS with failover & >> high > My failover is currently made by shell script which every 10 seconds > check for working FS and restarts it if it does not work. > I use svn trunk so crash happens once a while, but they are > successfully fixed by developers. > > Once there was a problem that conference module was stuck and did > not respond to my commands. I made script with netcat which > checks once a while for response and restarts if there is none. >> load. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/44622516/attachment-0002.html From rupa at rupa.com Thu Apr 2 05:37:28 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 2 Apr 2009 07:37:28 -0500 Subject: [Freeswitch-users] Use rates from lcr in nibblebill module In-Reply-To: <200904021322.05690.yivzhenko@mksat.net> References: <200904021322.05690.yivzhenko@mksat.net> Message-ID: Update the to the latest. I've added more channel vars: eg: after doing: (not a real number) I get the following: variable_lcr_query_digits: [12148267722] variable_lcr_query_profile: [0] variable_lcr_query_expanded_digits: [12148267722, 1214826772, 121482677, 12148267, 1214826, 121482, 12148, 1214, 121, 12, 1] variable_lcr_route_1: [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gateway/teliax/12148267722] variable_lcr_rate_1: [0.01000] variable_lcr_carrier_1: [teliax] variable_lcr_codec_1: [PCMU] variable_lcr_route_2: [[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec_string=PCMU]sofia/gateway/vitelity/12148267722] variable_lcr_rate_2: [0.01440] variable_lcr_carrier_2: [vitelity] variable_lcr_codec_2: [PCMU] variable_lcr_route_count: [2] variable_lcr_auto_route: [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gateway/teliax/12148267722|[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec_string=PCMU]sofia/gateway/vitelity/12148267722] variable_import: [lcr_carrier,lcr_rate] which, I think is what you are asking for. If you know which route you are going to use (eg: 1) then you can get it's rate by using lcr_rate_1. Alternatively, you can use the lcr_auto_route and then once the b-leg connects, query the b-leg variable for lcr_carrier and lcr_rate to see which one was actually used. You really can't use lcr_auto_route and set a single rate since each leg can be rated differently (look at example above). Normally lcr is used for your own rates between you and your carrier. That is independant of the rate table used for your customers. You can use lcr to query both. First use lcr to query your own rates using a different profile. This would return a *single* route if you setup your route table right. Save the rate in a var to be used with nibble bill. Then use lcr with your external rates so you can route according to your own cost with your carrier(s). On Thu, Apr 2, 2009 at 5:22 AM, Yuriy Ivzhenko (WP) wrote: > Hi, > > I want to use module lcr to find a best route and his rate , then make a > call > and bill on that rate with nibblebill module. > > lcr return variable "lcr_auto_route" that contains "[lcr_rate=xxx]" > variable > for new channel. > To use nibblebill i need to set "nibble_rate" = "lcr_rate". > > What is best method to do that? > Is there a way to do that with standard tools, without use external > scripts? > > > Thanks, > Yuriy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090402/bc23b3e1/attachment-0002.html From bipin at xbipin.com Thu Apr 2 06:01:57 2009 From: bipin at xbipin.com (xbipin) Date: Thu, 2 Apr 2009 06:01:57 -0700 (PDT) Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> Message-ID: <22847331.post@talk.nabble.com> hi, i wanted to know if there was any way to actually accept all registrations coming towards freeswitch, the normal function is to have all the suerid and passwords configured, but is there a way to accept all registrations coming towards a single ip or domain? Regards, Bipin -- View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Prometheus001 at gmx.net Thu Apr 2 06:08:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Apr 2009 15:08:25 +0200 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: <7C5053E3-1F30-4D6C-B786-004931E813A1@freeswitch.org> References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> <35b355e90904011759s43c91d4cg4d7a95b66d8027bd@mail.gmail.com> <7C5053E3-1F30-4D6C-B786-004931E813A1@freeswitch.org> Message-ID: <49D4B8C9.6070401@gmx.net> Wow, this is cool. Fantastic work! I tried this immediately. This is also very useful to share data across applications. Here an example how to share data between Freeswitch and a ruby memcache-client: On Ruby/Rails I set the namespace e.g. to "freeswitch" for the same memcached server in environment.rb In Freeswitch I add the following line to the dialplan: Take care to prefix your key (here "test") with the Ruby namespace "freeswitch:" Now you can receive the data in Ruby in raw mode: >> CACHE.get("test",0) => 'This is a test" The 0 as second parameter is important for the raw mode, otherwise ruby will try to marshall the result from memcached and fails. I added this info to the wiki. Best regards Peter Brian West schrieb: > At the very least you didn't say "I can't wait to play with it!" :P > > > On Apr 1, 2009, at 7:59 PM, Shelby Ramsey wrote: > >> Rupa, >> >> This is a big contribution! Thanks! Can't wait to play with this. >> >> SDR > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 leon at scarlet-internet.nl Thu Apr 2 06:13:01 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 2 Apr 2009 15:13:01 +0200 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <22847331.post@talk.nabble.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> Message-ID: <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> Hi, You can blindly accept registrations and / or authentication messages with these parameters in a sip profile: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#accept-blind-reg regards, Leon On Apr 2, 2009, at 3:01 PM, xbipin wrote: > > hi, > > i wanted to know if there was any way to actually accept all > registrations > coming towards freeswitch, the normal function is to have all the > suerid and > passwords configured, but is there a way to accept all registrations > coming > towards a single ip or domain? > > > Regards, > Bipin > -- > View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.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 rupa at rupa.com Thu Apr 2 06:13:23 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 2 Apr 2009 08:13:23 -0500 Subject: [Freeswitch-users] new module: mod_memcache In-Reply-To: <49D4B8C9.6070401@gmx.net> References: <191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com> <96CFC643-AB4E-4D57-877F-94CC10BB34EC@gmail.com> <35b355e90904011759s43c91d4cg4d7a95b66d8027bd@mail.gmail.com> <7C5053E3-1F30-4D6C-B786-004931E813A1@freeswitch.org> <49D4B8C9.6070401@gmx.net> Message-ID: On Thu, Apr 2, 2009 at 8:08 AM, Peter P GMX wrote: > Wow, this is cool. Fantastic work! > I tried this immediately. This is also very useful to share data across > applications. > [snip] > > I added this info to the wiki. > > Best regards > Peter > Thanks for the wiki update -- great to see examples of how to actually use it. :) -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/28e96090/attachment-0002.html From anthony.minessale at gmail.com Thu Apr 2 06:51:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Apr 2009 08:51:16 -0500 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: References: Message-ID: <191c3a030904020651ideeb958v77be91490a6ff38f@mail.gmail.com> Its the buffering and startup of the shout stream taking up the time, HINT put the shoutcast stream into a local stream with a .loc file and then play that in the conference. 2009/4/1 Rupa Schomaker > I've setup a conference bridge that has perpetual-sound set to a icecast > stream. When the first person connects, there is an ~7s delay before any > audio is heard. This is similar to a problem reported by Dan here and > concluded with Tony adding the channel var "enable_file_write_buffering". > The list discussion ended here: > http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011104.html > > > I set this var in my dialplan: > > > prior to joining the conference. > > The first person in still sees a 7s delay on audio the first time in. > > Like dan, I have icecast setup with > burst_on_connect set to 1 > but my burst_size is the default 64k so quite a bit of data. > > Has anyone been able to get an on-demand shoutcast stream from an icecast > server to start immediately (or at least within a second)? > > Thanks. > > -- > -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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/4a888955/attachment-0002.html From rupa at rupa.com Thu Apr 2 07:05:38 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 2 Apr 2009 09:05:38 -0500 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <191c3a030904020651ideeb958v77be91490a6ff38f@mail.gmail.com> References: <191c3a030904020651ideeb958v77be91490a6ff38f@mail.gmail.com> Message-ID: 2009/4/2 Anthony Minessale > Its the buffering and startup of the shout stream taking up the time, > > HINT put the shoutcast stream into a local stream with a .loc file and then > play that in the conference. > Ah, that is easy enough! Though I think with icecast doing the burst_on_connect thingie there should be enough data (pushed much faster than real time) to fill FS's buffers. But that would require mod_shout to cooperate with that strategy. ie: on connect, drain the socket as fast as it can filling it's own buffers. Once it's own buffers are full start streaming. I think right now it drains the socket only as fast as it needs to. Or maybe not. I'll go the local stream route for now.... -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/0372a030/attachment-0002.html From Prometheus001 at gmx.net Thu Apr 2 07:05:51 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Apr 2009 16:05:51 +0200 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> Message-ID: <49D4C63F.8050400@gmx.net> I use the access control list acl.conf.xml to configure that. Put ip/mask into the domain part of this config file, then it accepts calls from this ip (eg. ip/32.) or from this subnet (e.g. ip/24). Best regards Peter Leon de Rooij schrieb: > Hi, > > You can blindly accept registrations and / or authentication messages > with these parameters in a sip profile: > > > > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#accept-blind-reg > > regards, > > Leon > > On Apr 2, 2009, at 3:01 PM, xbipin wrote: > > >> hi, >> >> i wanted to know if there was any way to actually accept all >> registrations >> coming towards freeswitch, the normal function is to have all the >> suerid and >> passwords configured, but is there a way to accept all registrations >> coming >> towards a single ip or domain? >> >> >> Regards, >> Bipin >> -- >> View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.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 >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mike at jerris.com Thu Apr 2 07:07:53 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 2 Apr 2009 10:07:53 -0400 Subject: [Freeswitch-users] about freeswitch conference In-Reply-To: <200904021529137966712@gmail.com> References: <200904021524116567464@gmail.com> <200904021529137966712@gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Mod_conference On Apr 2, 2009, at 3:29 AM, bmsword wrote: > I want to use another softswitch conference that has been > deployed in freeswitch,How should I do? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/804ee06f/attachment-0002.html From bipin at xbipin.com Thu Apr 2 07:38:35 2009 From: bipin at xbipin.com (Bipin Patel) Date: Thu, 02 Apr 2009 18:38:35 +0400 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <49D4C63F.8050400@gmx.net> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> Message-ID: <49D4CDEB.1040201@xbipin.com> hi, will the below work if all the registration that is to be accepted come from different public ip addresses, i mean, clients from all ip ranges and addresses rather than a single ip Regards, Bipin www.xbipin.com +971-55-9270058 -------- Original Message -------- Subject: Re: [Freeswitch-users] upper registration in FS? From: Peter P GMX To: freeswitch-users at lists.freeswitch.org Date: Thursday, April 02, 2009 6:05:51 PM > I use the access control list acl.conf.xml to configure that. > > Put ip/mask into the domain part of this config file, then it accepts > calls from this ip (eg. ip/32.) or from this subnet (e.g. ip/24). > > > > > > > > Best regards > Peter > > Leon de Rooij schrieb: >> Hi, >> >> You can blindly accept registrations and / or authentication messages >> with these parameters in a sip profile: >> >> >> >> >> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#accept-blind-reg >> >> regards, >> >> Leon >> >> On Apr 2, 2009, at 3:01 PM, xbipin wrote: >> >> >>> hi, >>> >>> i wanted to know if there was any way to actually accept all >>> registrations >>> coming towards freeswitch, the normal function is to have all the >>> suerid and >>> passwords configured, but is there a way to accept all registrations >>> coming >>> towards a single ip or domain? >>> >>> >>> Regards, >>> Bipin >>> -- >>> View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.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 >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > __________ NOD32 3983 (20090402) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > > > From intralanman at freeswitch.org Thu Apr 2 07:50:21 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 02 Apr 2009 10:50:21 -0400 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <49D4CDEB.1040201@xbipin.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> <49D4CDEB.1040201@xbipin.com> Message-ID: <49D4D0AD.9030904@freeswitch.org> Bipin Patel wrote: > hi, > > will the below work if all the registration that is to be accepted come > from different public ip addresses, i mean, clients from all ip ranges > and addresses rather than a single ip > yeah, that's kinda why its called "blind" ... you don't have to know where its coming from, and it doesn't have to be valid... just "blindly" accepts it -Ray From Richard.Lamkin at mettoni.com Thu Apr 2 08:12:22 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 2 Apr 2009 16:12:22 +0100 Subject: [Freeswitch-users] Database schema Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804BE74B1@nickel.mettonigroup.com> Are there documents or wiki page [I've missed during my searches] that detail the records and their types that are stored in the various FS databases; e.g. sofia_reg_.db, core.db ? Regards Richard Lamkin ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/f99c8aa5/attachment-0002.html From brian at freeswitch.org Thu Apr 2 08:21:44 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 10:21:44 -0500 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <49D4CDEB.1040201@xbipin.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> <49D4CDEB.1040201@xbipin.com> Message-ID: <27317F98-DDB5-49D3-93B8-F0B8949710FB@freeswitch.org> Turn on Multireg too. /b On Apr 2, 2009, at 9:38 AM, Bipin Patel wrote: > hi, > > will the below work if all the registration that is to be accepted > come > from different public ip addresses, i mean, clients from all ip ranges > and addresses rather than a single ip > > > > > Regards, > Bipin > www.xbipin.com > +971-55-9270058 Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/9245e267/attachment-0002.html From mike at jerris.com Thu Apr 2 08:28:32 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 2 Apr 2009 11:28:32 -0400 Subject: [Freeswitch-users] Database schema In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804BE74B1@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804BE74B1@nickel.mettonigroup.com> Message-ID: <6C26E3C4-1230-4BE7-A6DD-A5B4ECADBD95@jerris.com> no, but they all auto-create. You can create a db and set up odbc, start freeswitch, then dump your db schema. Also, please do not send confidential emails to the mailing list. Mike On Apr 2, 2009, at 11:12 AM, Richard Lamkin wrote: > Are there documents or wiki page [I?ve missed during my searches] > that detail the records and their types that are stored in the > various FS databases; e.g. sofia_reg_.db, core.db ? > > Regards > Richard Lamkin > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/b5cb06b2/attachment-0002.html From bipin at xbipin.com Thu Apr 2 08:40:14 2009 From: bipin at xbipin.com (Bipin Patel) Date: Thu, 02 Apr 2009 19:40:14 +0400 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <27317F98-DDB5-49D3-93B8-F0B8949710FB@freeswitch.org> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> <49D4CDEB.1040201@xbipin.com> <27317F98-DDB5-49D3-93B8-F0B8949710FB@freeswitch.org> Message-ID: <49D4DC5E.4080506@xbipin.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/ff02f04e/attachment-0002.html From cstomi.levlist at gmail.com Thu Apr 2 09:46:57 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Thu, 02 Apr 2009 18:46:57 +0200 Subject: [Freeswitch-users] loopback-b channels stay alive Message-ID: <49D4EC01.6050205@gmail.com> Hello, We originate loopback channels and they end up in calling sofia and transfer the call to a fifo. If we have a heavy call volume loopback-b channels don't hangup properly. They stay in core.db. Unfortunetly we can't reproduce it on test boxes but happens every day. On this box we had to turn off debug logging, becase we had I/O problems. The only thing I saw in log that switch_core_session_thread don't call switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n", session->id, switch_channel_get_name(session->channel)); in these cases. We have local patches (I don't think they are related) and we are running FS on virtual machine and we had some problem with that before so I'm not sure, but I guess it is maybe a lock or mutex problem. I tried SWITCH_DEBUG_RWLOCKS, but I got build error, and I don't know what to do with it. FS_CFLAGS = -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS export CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" export MOD_CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" ./configure gcc -I/DEVEL/freeswitch/src/include -I/DEVEL/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS -Wall -std=c99 -pedantic -o .libs/freeswitch freeswitch-switch.o -lm ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath -Wl,/opt/freeswitch//lib ./.libs/libfreeswitch.so: undefined reference to `switch_core_session_read_lock' ./.libs/libfreeswitch.so: undefined reference to `switch_core_session_locate' ./.libs/libfreeswitch.so: undefined reference to `switch_core_session_rwunlock' collect2: ld returned 1 exit status make[2]: *** [freeswitch] Error 1 Could you please tell me how could I test mutexes, rwlocks? Other option would be to omit loopback channels. Anthony earlier suggested me to avoid it and call sofia directly "you could make the loopback channel execute the eval app and do the originate to the sofia channel from the dialplan. or make the loopback chan exec a lua or js and fire an originate command and exit This way you don't have the loopback a and b leg as well as the sofia chan." but it doesn't work, because originate api doesn't let us originate inside a session. So we still using it. Thanks in advance, Tamas From msc at freeswitch.org Thu Apr 2 10:07:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 2 Apr 2009 10:07:18 -0700 Subject: [Freeswitch-users] loopback-b channels stay alive In-Reply-To: <49D4EC01.6050205@gmail.com> References: <49D4EC01.6050205@gmail.com> Message-ID: <87f2f3b90904021007j1d2ae759n388d05078c826219@mail.gmail.com> Thanks for doing some of the legwork on this. BTW, this thread is probably a bit too technical for the users list - I recommend sending to the dev list. :) -MC On Thu, Apr 2, 2009 at 9:46 AM, Tamas Cseke wrote: > Hello, > > We originate loopback channels and they end up in calling sofia > and transfer the call to a fifo. > > If we have a heavy call volume loopback-b channels don't hangup properly. > They stay in core.db. > Unfortunetly we can't reproduce it on test boxes but happens every day. > On this box we had to turn off debug logging, becase we had I/O problems. > > The only thing I saw in log that switch_core_session_thread don't call > > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Session %" > SWITCH_SIZE_T_FMT " (%s) Ended\n", > session->id, > switch_channel_get_name(session->channel)); > > in these cases. > We have local patches (I don't think they are related) and we are > running FS on virtual machine and we had some problem with that before > so I'm not sure, but I guess it is maybe a lock or mutex problem. > > I tried SWITCH_DEBUG_RWLOCKS, but I got build error, and I don't know > what to do with it. > > FS_CFLAGS = -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS > export CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" > export MOD_CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" > ./configure > > gcc -I/DEVEL/freeswitch/src/include > -I/DEVEL/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS -Wall -std=c99 > -pedantic -o .libs/freeswitch freeswitch-switch.o -lm > ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt > -lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath > -Wl,/opt/freeswitch//lib > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_read_lock' > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_locate' > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_rwunlock' > collect2: ld returned 1 exit status > make[2]: *** [freeswitch] Error 1 > > Could you please tell me how could I test mutexes, rwlocks? > > Other option would be to omit loopback channels. > Anthony earlier suggested me to avoid it and call sofia directly > > "you could make the loopback channel execute the eval app and do the > originate to the sofia channel from the dialplan. > > > or make the loopback chan exec a lua or js and fire an originate command > and > exit > > This way you don't have the loopback a and b leg as well as the sofia > chan." > > but it doesn't work, because originate api doesn't let us originate inside > a session. > So we still using it. > > > Thanks in advance, > Tamas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/3e1381c5/attachment-0002.html From brian at freeswitch.org Thu Apr 2 10:20:31 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 12:20:31 -0500 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <2d9149cd0904012309h5428ece5qce9a287b0972dd81@mail.gmail.com> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <8b3b7acc0904011641v2c7e3768ne72e06623ed064f2@mail.gmail.com> <2d9149cd0904012309h5428ece5qce9a287b0972dd81@mail.gmail.com> Message-ID: hehe I emailed it to him off list :) /b On Apr 2, 2009, at 1:09 AM, Kristian Kielhofner wrote: > I probably shouldn't be doing this for you, but... > > http://bugs.digium.com/view.php?id=14431 > > ;) Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/c8a23c63/attachment-0002.html From anthony.minessale at gmail.com Thu Apr 2 11:03:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Apr 2009 13:03:51 -0500 Subject: [Freeswitch-users] loopback-b channels stay alive In-Reply-To: <49D4EC01.6050205@gmail.com> References: <49D4EC01.6050205@gmail.com> Message-ID: <191c3a030904021103o40307040xfd5489763644ea72@mail.gmail.com> you can't pass it in with -D you have to actually add #define SWITCH_DEBUG_RWLOCKS to the top of switch_core.h On Thu, Apr 2, 2009 at 11:46 AM, Tamas Cseke wrote: > Hello, > > We originate loopback channels and they end up in calling sofia > and transfer the call to a fifo. > > If we have a heavy call volume loopback-b channels don't hangup properly. > They stay in core.db. > Unfortunetly we can't reproduce it on test boxes but happens every day. > On this box we had to turn off debug logging, becase we had I/O problems. > > The only thing I saw in log that switch_core_session_thread don't call > > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Session %" > SWITCH_SIZE_T_FMT " (%s) Ended\n", > session->id, > switch_channel_get_name(session->channel)); > > in these cases. > We have local patches (I don't think they are related) and we are > running FS on virtual machine and we had some problem with that before > so I'm not sure, but I guess it is maybe a lock or mutex problem. > > I tried SWITCH_DEBUG_RWLOCKS, but I got build error, and I don't know > what to do with it. > > FS_CFLAGS = -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS > export CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" > export MOD_CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" > ./configure > > gcc -I/DEVEL/freeswitch/src/include > -I/DEVEL/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS -Wall -std=c99 > -pedantic -o .libs/freeswitch freeswitch-switch.o -lm > ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt > -lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath > -Wl,/opt/freeswitch//lib > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_read_lock' > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_locate' > ./.libs/libfreeswitch.so: undefined reference to > `switch_core_session_rwunlock' > collect2: ld returned 1 exit status > make[2]: *** [freeswitch] Error 1 > > Could you please tell me how could I test mutexes, rwlocks? > > Other option would be to omit loopback channels. > Anthony earlier suggested me to avoid it and call sofia directly > > "you could make the loopback channel execute the eval app and do the > originate to the sofia channel from the dialplan. > > > or make the loopback chan exec a lua or js and fire an originate command > and > exit > > This way you don't have the loopback a and b leg as well as the sofia > chan." > > but it doesn't work, because originate api doesn't let us originate inside > a session. > So we still using it. > > > Thanks in advance, > Tamas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/3090d55b/attachment-0002.html From solko at gcdf.pl Thu Apr 2 12:29:28 2009 From: solko at gcdf.pl (Szymon Olko) Date: Thu, 02 Apr 2009 21:29:28 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> Message-ID: <49D51218.2080209@gcdf.pl> Brian West pisze: > > On Apr 2, 2009, at 4:13 AM, Szymon Olko wrote: > >> I did not think yet about HA nor LB. >> >> I tested how FS handles high load. All my calls are placed in >> mod_conference. When cpu usage gets it's limits then new calls can >> be placed but sound quality is getting worst with every next call. >> When calls are hanged up then sound gets better. I did not test >> it to see what happens when more and more calls are created. >> FS has very low memory consumption and I think that CPU is the limit. >> I did not notice any monitoring of CPU usage by FS, but my >> installation is limited to only few modules, so maybe I'm missing >> something. > > Load testing against the conference module is about the worst thing you > can do... tossing 100+ people in the same conference isn't going to > scale well for load testing because its not something you usually do in > a real world scenario. Usually you'll have most of the participants muted. > > I highly recommend you try doing something like a bridge or a file > playback from a ram disk. > I did not described it perfectly. I made agents, queues scenarios on conferences. This what I tested was for example 100 calls, so it's 200 channels, and 100 conferences, 2 channels per conference, all are unmuted. I did that just because it is my work scenario. >>> >>> Would love to hear some experiences of deploying FS with failover & high >> My failover is currently made by shell script which every 10 seconds >> check for working FS and restarts it if it does not work. >> I use svn trunk so crash happens once a while, but they are >> successfully fixed by developers. >> >> Once there was a problem that conference module was stuck and did not >> respond to my commands. I made script with netcat which >> checks once a while for response and restarts if there is none. >>> load. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 2 12:34:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 14:34:51 -0500 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <49D51218.2080209@gcdf.pl> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> <49D51218.2080209@gcdf.pl> Message-ID: <2CD5CE30-E2E5-4D72-A587-19F92D862665@freeswitch.org> what kind of hardware? /b On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: > I did not described it perfectly. I made agents, queues scenarios on > conferences. > This what I tested was for example 100 calls, so it's 200 channels, > and 100 conferences, 2 channels per conference, all are > unmuted. I did that just because it is my work scenario. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/c6244405/attachment-0002.html From Prometheus001 at gmx.net Thu Apr 2 13:07:10 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Apr 2009 22:07:10 +0200 Subject: [Freeswitch-users] Handle invite with wrong to:IP Message-ID: <49D51AEE.7010904@gmx.net> Hello, I am using a SIP account from Netvoip CH. I try to receive inbound call from this SIP trunk. I discovered that, when they sent an invite, the IP-Adress of the to: is their own IP address. There fore ACL doesn't work and FS asks for authorization, which then fails I receive the following message on the CLI: 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() Can't find user [anonymous at 62.65.128.62] You must define a domain called '62.65.128.62' in your directory and add a user with the id="anonymous" attribute and you must configure your device to use the proper domain in it's authentication credentials. I could do that, but this is not clean and I do not have a password for that. How can I workaround this, so that Freeswitch accepts this call? Aliases do not seem to work. Here is a sample message after FS asks for authorization: xx.xx.xxx.xxx is the IP of our Freeswitch 62.65.128.62 is the IP of Netvoip CH I would expect To: . instead of To: . U 62.65.128.62:5060 -> xx.xx.xxx.xxx:5080 INVITE sip:0715aaaaaa at xx.xx.xxx.xxx:5080 SIP/2.0. Via: SIP/2.0/UDP 62.65.128.62:5060. Via: SIP/2.0/UDP 62.65.128.61:5060;branch=z9hG4bK8c977d2613c4d7d1fd9d03d4. Max-Forwards: 69. From: ;tag=8c977d2613672832fd9d03e9. To: . Call-ID: 8c977d261329cd80fd9d03d6 at 62.65.128.61. CSeq: 2 INVITE. User-agent: Netstream VoIP Gateway. Contact: . Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 584. Proxy-Authorization: Digest username="anonymous", realm="62.65.128.62", nonce="a4151ee0-1fbb-11de-b056-494b9de21e06", nc="00000001", uri="sip:0715aaaaaa at 62.65.128.62:5060", cnonce="5f109eee", response="62faa6d38b3b12c3626753395a8b507c", algorithm="MD5", qop="auth". . v=0. o=- 225947743692042 1 IN IP4 62.65.128.62. s=-. c=IN IP4 62.65.128.62. t=0 0. m=audio 28224 RTP/AVP 8 18 4 3 100 100 99 100 100 98 97 96 105 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=no. a=rtpmap:3 GSM/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:99 G726-16/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:97 G726-32/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:105 iLBC/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. Best regards Peter From solko at gcdf.pl Thu Apr 2 13:07:53 2009 From: solko at gcdf.pl (Szymon Olko) Date: Thu, 02 Apr 2009 22:07:53 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <2CD5CE30-E2E5-4D72-A587-19F92D862665@freeswitch.org> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> <49D51218.2080209@gcdf.pl> <2CD5CE30-E2E5-4D72-A587-19F92D862665@freeswitch.org> Message-ID: <49D51B19.3050709@gcdf.pl> Brian West pisze: > what kind of hardware? > I made testes on Pentium-M laptop with single core 1,6Hz. I did not write those results, it was over 100 calls that was handle good, I was just curios what will happen. Tomorrow I will make real testes. My production works on 2 core P4 and I have there only 35 agents CPU load is like 7% with 15% small peeks. All phones are sip or analog via sip gateways, PRI is currently still on asterisk which is connected via sip. > /b > > On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: > >> I did not described it perfectly. I made agents, queues scenarios on >> conferences. >> This what I tested was for example 100 calls, so it's 200 channels, >> and 100 conferences, 2 channels per conference, all are >> unmuted. I did that just because it is my work scenario. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 ceino.no at gmail.com Thu Apr 2 12:55:25 2009 From: ceino.no at gmail.com (Ceino) Date: Thu, 02 Apr 2009 21:55:25 +0200 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre3 Now Available In-Reply-To: <87f2f3b90904010949x4d3fc2f8ia9feda2581956f68@mail.gmail.com> References: <87f2f3b90904010949x4d3fc2f8ia9feda2581956f68@mail.gmail.com> Message-ID: <49D5182D.1080508@gmail.com> Hi, I have tested it a little bit and it's works well. But when I give it the command to stop (...) it use about 40 sec. to stop (1.0.3 use about 5 sec). Here is a log over where is hang (looks like a Sofia thread use long time to stop): ------------------------------------------------------------------ 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal-ipv6 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 sofia_glue_del_profile() deleted gateway example.com 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock external ------------------------------------------------------------------ Best Regards Lars Sivertsen Michael Collins wrote: > The FreeSWITCH team would like to let everyone know that the latest > version is available. More information can be found here: > http://www.freeswitch.org/node/172 > > By all means download, upgrade, test, and report back! Your feedback > helps us make FreeSWITCH even better! > -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 > From ceino.no at gmail.com Thu Apr 2 13:04:28 2009 From: ceino.no at gmail.com (Ceino) Date: Thu, 02 Apr 2009 22:04:28 +0200 Subject: [Freeswitch-users] FreeSWITCH 1.0.4pre3 is slow to stop Message-ID: <49D51A4C.7040701@gmail.com> Hi all, I have tested it a little bit and it's works well. But when I give it the command to stop (...) it use about 40 sec to stop (1.0.3 use about 5 sec). Here is a log over where is hang (looks like a Sofia thread use long time to stop): ------------------------------------------------------------------ 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal-ipv6 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 sofia_glue_del_profile() deleted gateway example.com 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock external ------------------------------------------------------------------ Best Regards Lars Sivertsen From brian at freeswitch.org Thu Apr 2 13:11:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 15:11:22 -0500 Subject: [Freeswitch-users] FreeSWITCH 1.0.4pre3 is slow to stop In-Reply-To: <49D51A4C.7040701@gmail.com> References: <49D51A4C.7040701@gmail.com> Message-ID: <73F10AE0-6EDA-4F02-A4D9-BA8AF73CB070@freeswitch.org> Try updating to SVN trunk... I think we fixed that already. /b On Apr 2, 2009, at 3:04 PM, Ceino wrote: > Hi all, I have tested it a little bit and it's works well. But when I > give it the command to stop (...) > it use about 40 sec to stop (1.0.3 use about 5 sec). > > Here is a log over where is hang (looks like a Sofia thread use long > time to stop): > ------------------------------------------------------------------ > > 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() > Waiting for worker thread > 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() > Write > unlock internal-ipv6 > 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() > Write > unlock internal > 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() > Waiting for worker thread > 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 > sofia_glue_del_profile() > deleted gateway example.com > 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() > Write > unlock external > ------------------------------------------------------------------ > > > Best Regards > > Lars Sivertsen Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/4283c560/attachment-0002.html From brian at freeswitch.org Thu Apr 2 13:14:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Apr 2009 15:14:42 -0500 Subject: [Freeswitch-users] Handle invite with wrong to:IP In-Reply-To: <49D51AEE.7010904@gmx.net> References: <49D51AEE.7010904@gmx.net> Message-ID: We use the true network ip the invite came from NOT the one in the sip headers. Not very trust worth to do that you think? ;) So if your ACL is correctly setup to 62.65.128.62 it would let them in please verify your ACL is correct... /b On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote: > Hello, > > I am using a SIP account from Netvoip CH. I try to receive inbound > call > from this SIP trunk. I discovered that, when they sent an invite, the > IP-Adress of the to: is their own IP address. > There fore ACL doesn't work and FS asks for authorization, which > then fails > > I receive the following message on the CLI: > 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() > Can't find user [anonymous at 62.65.128.62] > You must define a domain called '62.65.128.62' in your directory and > add > a user with the id="anonymous" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/5fa2980e/attachment-0002.html From anthony.minessale at gmail.com Thu Apr 2 13:14:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Apr 2009 15:14:53 -0500 Subject: [Freeswitch-users] Handle invite with wrong to:IP In-Reply-To: <49D51AEE.7010904@gmx.net> References: <49D51AEE.7010904@gmx.net> Message-ID: <191c3a030904021314o461ef854hcf856be9f406f38e@mail.gmail.com> acl uses the remote addr from the socket connection, not anything from the sip message. On Thu, Apr 2, 2009 at 3:07 PM, Peter P GMX wrote: > Hello, > > I am using a SIP account from Netvoip CH. I try to receive inbound call > from this SIP trunk. I discovered that, when they sent an invite, the > IP-Adress of the to: is their own IP address. > There fore ACL doesn't work and FS asks for authorization, which then fails > > I receive the following message on the CLI: > 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() > Can't find user [anonymous at 62.65.128.62] > You must define a domain called '62.65.128.62' in your directory and add > a user with the id="anonymous" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > > I could do that, but this is not clean and I do not have a password for > that. > > How can I workaround this, so that Freeswitch accepts this call? Aliases > do not seem to work. > > Here is a sample message after FS asks for authorization: > xx.xx.xxx.xxx is the IP of our Freeswitch > 62.65.128.62 is the IP of Netvoip CH > > I would expect > To: . > instead of > To: >. > > U 62.65.128.62:5060 -> xx.xx.xxx.xxx:5080 > INVITE sip:0715aaaaaa at xx.xx.xxx.xxx:5080 SIP/2.0. > Via: SIP/2.0/UDP 62.65.128.62:5060. > Via: SIP/2.0/UDP 62.65.128.61:5060;branch=z9hG4bK8c977d2613c4d7d1fd9d03d4. > Max-Forwards: 69. > From: > >;tag=8c977d2613672832fd9d03e9. > To: >. > Call-ID: 8c977d261329cd80fd9d03d6 at 62.65.128.61. > CSeq: 2 INVITE. > User-agent: Netstream VoIP Gateway. > Contact: . > Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE. > Content-Type: application/sdp. > Content-Length: 584. > Proxy-Authorization: Digest username="anonymous", realm="62.65.128.62", > nonce="a4151ee0-1fbb-11de-b056-494b9de21e06", nc="00000001", > uri="sip:0715aaaaaa at 62.65.128.62:5060", cnonce="5f109eee", > response="62faa6d38b3b12c3626753395a8b507c", algorithm="MD5", qop="auth". > . > v=0. > o=- 225947743692042 1 IN IP4 62.65.128.62. > s=-. > c=IN IP4 62.65.128.62. > t=0 0. > m=audio 28224 RTP/AVP 8 18 4 3 100 100 99 100 100 98 97 96 105 0 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:4 G723/8000. > a=fmtp:4 annexa=no. > a=rtpmap:3 GSM/8000. > a=rtpmap:100 speex/8000. > a=rtpmap:100 speex/8000. > a=rtpmap:99 G726-16/8000. > a=rtpmap:100 speex/8000. > a=rtpmap:100 speex/8000. > a=rtpmap:98 G726-24/8000. > a=rtpmap:97 G726-32/8000. > a=rtpmap:96 G726-40/8000. > a=rtpmap:105 iLBC/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/fbea3fad/attachment-0002.html From anthony.minessale at gmail.com Thu Apr 2 13:15:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Apr 2009 15:15:58 -0500 Subject: [Freeswitch-users] FreeSWITCH 1.0.4pre3 is slow to stop In-Reply-To: <49D51A4C.7040701@gmail.com> References: <49D51A4C.7040701@gmail.com> Message-ID: <191c3a030904021315m130ab671t9e0c94f4bf7973e9@mail.gmail.com> wait for pre4 On Thu, Apr 2, 2009 at 3:04 PM, Ceino wrote: > Hi all, I have tested it a little bit and it's works well. But when I > give it the command to stop (...) > it use about 40 sec to stop (1.0.3 use about 5 sec). > > Here is a log over where is hang (looks like a Sofia thread use long > time to stop): > ------------------------------------------------------------------ > > 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() > Waiting for worker thread > 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write > unlock internal-ipv6 > 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write > unlock internal > 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() > Waiting for worker thread > 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 sofia_glue_del_profile() > deleted gateway example.com > 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write > unlock external > ------------------------------------------------------------------ > > > Best Regards > > Lars Sivertsen > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/89631066/attachment-0002.html From Prometheus001 at gmx.net Thu Apr 2 13:34:10 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Apr 2009 22:34:10 +0200 Subject: [Freeswitch-users] Handle invite with wrong to:IP In-Reply-To: References: <49D51AEE.7010904@gmx.net> Message-ID: <49D52142.7040401@gmx.net> My ACL contains: So this should be fine, right? However it doesn't work. Best regards Peter Brian West schrieb: > We use the true network ip the invite came from NOT the one in the sip > headers. Not very trust worth to do that you think? ;) > > So if your ACL is correctly setup to 62.65.128.62 it would let them in > please verify your ACL is correct... > > /b > > On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote: > >> Hello, >> >> I am using a SIP account from Netvoip CH. I try to receive inbound call >> from this SIP trunk. I discovered that, when they sent an invite, the >> IP-Adress of the to: is their own IP address. >> There fore ACL doesn't work and FS asks for authorization, which then >> fails >> >> I receive the following message on the CLI: >> 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() >> Can't find user [anonymous at 62.65.128.62 ] >> You must define a domain called '62.65.128.62' in your directory and add >> a user with the id="anonymous" attribute >> and you must configure your device to use the proper domain in it's >> authentication credentials. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 2 14:24:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Apr 2009 16:24:29 -0500 Subject: [Freeswitch-users] Handle invite with wrong to:IP In-Reply-To: <49D52142.7040401@gmx.net> References: <49D51AEE.7010904@gmx.net> <49D52142.7040401@gmx.net> Message-ID: <191c3a030904021424ta0f4adwfe805f9d9bda86e@mail.gmail.com> look at the debug log and see what happens? On Thu, Apr 2, 2009 at 3:34 PM, Peter P GMX wrote: > My ACL contains: > > > > > > So this should be fine, right? However it doesn't work. > > Best regards > Peter > > > Brian West schrieb: > > We use the true network ip the invite came from NOT the one in the sip > > headers. Not very trust worth to do that you think? ;) > > > > So if your ACL is correctly setup to 62.65.128.62 it would let them in > > please verify your ACL is correct... > > > > /b > > > > On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote: > > > >> Hello, > >> > >> I am using a SIP account from Netvoip CH. I try to receive inbound call > >> from this SIP trunk. I discovered that, when they sent an invite, the > >> IP-Adress of the to: is their own IP address. > >> There fore ACL doesn't work and FS asks for authorization, which then > >> fails > >> > >> I receive the following message on the CLI: > >> 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() > >> Can't find user [anonymous at 62.65.128.62 >] > >> You must define a domain called '62.65.128.62' in your directory and add > >> a user with the id="anonymous" attribute > >> and you must configure your device to use the proper domain in it's > >> authentication credentials. > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/c2902cb0/attachment-0002.html From Prometheus001 at gmx.net Thu Apr 2 14:45:40 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Apr 2009 23:45:40 +0200 Subject: [Freeswitch-users] Handle invite with wrong to:IP In-Reply-To: <191c3a030904021424ta0f4adwfe805f9d9bda86e@mail.gmail.com> References: <49D51AEE.7010904@gmx.net> <49D52142.7040401@gmx.net> <191c3a030904021424ta0f4adwfe805f9d9bda86e@mail.gmail.com> Message-ID: <49D53204.3090701@gmx.net> I restart FS and initiate an incoming call (trunk is registered at the SIP provider). This is what I see on the console: . . . 2009-04-02 23:39:16 [DEBUG] mod_event_socket.c:2224 mod_event_socket_runtime() Socket up listening on 0.0.0.0:8021 2009-04-02 23:39:16 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding xxx.xxx.xxx.xxx/32 (allow) to list strict 2009-04-02 23:39:16 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding xx.xx.xxx.xx/32 (allow) to list domains 2009-04-02 23:39:16 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 62.65.128.62/32 (allow) to list domains 2009-04-02 23:39:48 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() Can't find user [anonymous at 62.65.128.62] You must define a domain called '62.65.128.62' in your directory and add a user with the id="anonymous" attribute and you must configure your device to use the proper domain in it's authentication credentials. Nothing else. Here is the registration info: Name Netvoip Scheme Digest Realm sip.netvoip.ch Username 071xxxxxxx Password yes From Contact Exten 071xxxxxxx To sip:071xxxxxxx at sip.netvoip.ch Proxy sip:sip.netvoip.ch Context public Expires 60 Freq 60 Ping 0 PingFreq 0 State REGED Status UP CallsIN 0 CallsOUT 0 Best regards Peter Anthony Minessale schrieb: > look at the debug log and see what happens? > > On Thu, Apr 2, 2009 at 3:34 PM, Peter P GMX > wrote: > > My ACL contains: > > > > > > So this should be fine, right? However it doesn't work. > > Best regards > Peter > > > Brian West schrieb: > > We use the true network ip the invite came from NOT the one in > the sip > > headers. Not very trust worth to do that you think? ;) > > > > So if your ACL is correctly setup to 62.65.128.62 it would let > them in > > please verify your ACL is correct... > > > > /b > > > > On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote: > > > >> Hello, > >> > >> I am using a SIP account from Netvoip CH. I try to receive > inbound call > >> from this SIP trunk. I discovered that, when they sent an > invite, the > >> IP-Adress of the to: is their own IP address. > >> There fore ACL doesn't work and FS asks for authorization, > which then > >> fails > >> > >> I receive the following message on the CLI: > >> 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 > sofia_reg_parse_auth() > >> Can't find user [anonymous at 62.65.128.62 > >] > >> You must define a domain called '62.65.128.62' in your > directory and add > >> a user with the id="anonymous" attribute > >> and you must configure your device to use the proper domain in it's > >> authentication credentials. > > > > Brian West > > brian at freeswitch.org > > > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > 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/ > > 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 red.rain.seven at gmail.com Thu Apr 2 15:26:38 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Thu, 2 Apr 2009 15:26:38 -0700 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <49D51B19.3050709@gcdf.pl> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> <49D51218.2080209@gcdf.pl> <2CD5CE30-E2E5-4D72-A587-19F92D862665@freeswitch.org> <49D51B19.3050709@gcdf.pl> Message-ID: <59ad9ca10904021526s539417b1g7f5081f927ccf77@mail.gmail.com> How do you load balance conference calls? Doesn't all the conference members have to be on the same freeswitch server? On Thu, Apr 2, 2009 at 1:07 PM, Szymon Olko wrote: > Brian West pisze: > > what kind of hardware? > > > I made testes on Pentium-M laptop with single core 1,6Hz. I did not write > those results, it was over 100 calls that was handle > good, I was just curios what will happen. Tomorrow I will make real testes. > My production works on 2 core P4 and I have there only > 35 agents CPU load is like 7% with 15% small peeks. > > All phones are sip or analog via sip gateways, PRI is currently still on > asterisk which is connected via sip. > > > /b > > > > On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: > > > >> I did not described it perfectly. I made agents, queues scenarios on > >> conferences. > >> This what I tested was for example 100 calls, so it's 200 channels, > >> and 100 conferences, 2 channels per conference, all are > >> unmuted. I did that just because it is my work scenario. > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090402/b70e45ab/attachment-0002.html From bipin at xbipin.com Thu Apr 2 22:59:57 2009 From: bipin at xbipin.com (xbipin) Date: Thu, 2 Apr 2009 22:59:57 -0700 (PDT) Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <49D4DC5E.4080506@xbipin.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> <49D4CDEB.1040201@xbipin.com> <27317F98-DDB5-49D3-93B8-F0B8949710FB@freeswitch.org> <49D4DC5E.4080506@xbipin.com> Message-ID: <22862459.post@talk.nabble.com> hi, any1 have any idea how what to sue in dialplan such that calls from a single id go to a specific gateway only with blind registration enabled, this is the only major issue im having. Regards, Bipin -- View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22862459.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Thu Apr 2 23:53:35 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 3 Apr 2009 17:53:35 +1100 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <22862459.post@talk.nabble.com> References: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> <22847331.post@talk.nabble.com> <8C731DA3-E740-479E-ADC8-FB04E4119A0C@scarlet-internet.nl> <49D4C63F.8050400@gmx.net> <49D4CDEB.1040201@xbipin.com> <27317F98-DDB5-49D3-93B8-F0B8949710FB@freeswitch.org> <49D4DC5E.4080506@xbipin.com> <22862459.post@talk.nabble.com> Message-ID: <20090403065335.GA5645@jdc.jasonjgw.net> xbipin wrote: > > any1 have any idea how what to sue in dialplan such that calls from a single > id go to a specific gateway only with blind registration enabled, this is > the only major issue im having. Perhaps you could match the source address in the dial plan and then bridge or redirect the call to the desired gateway. for example. I tested a similar example once and it did work. From solko at gcdf.pl Fri Apr 3 00:27:23 2009 From: solko at gcdf.pl (Szymon Olko) Date: Fri, 03 Apr 2009 09:27:23 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <59ad9ca10904021526s539417b1g7f5081f927ccf77@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <49D481CC.70102@gcdf.pl> <70EE24A7-F6C1-49F5-B212-8DD17F51EAA4@freeswitch.org> <49D51218.2080209@gcdf.pl> <2CD5CE30-E2E5-4D72-A587-19F92D862665@freeswitch.org> <49D51B19.3050709@gcdf.pl> <59ad9ca10904021526s539417b1g7f5081f927ccf77@mail.gmail.com> Message-ID: <49D5BA5B.5070104@gcdf.pl> Henry Huang pisze: > How do you load balance conference calls? Doesn't all the conference > members have to be on the same freeswitch server? > As I wrote I do not load balance them yet. I didn't investigate that but what comes to my mind is to setup 2 FS end register agents to one of them (load balance them), sip phones through proxy server. Then one separate FS for incoming calls and in that FS place my queue system. When incoming call needs to be connected to agent then right FS machine would be choosen. This just idea I believe that in time I will need something like that FS developers will give us some modules or other options. > On Thu, Apr 2, 2009 at 1:07 PM, Szymon Olko wrote: > > Brian West pisze: > > what kind of hardware? > > > I made testes on Pentium-M laptop with single core 1,6Hz. I did not > write those results, it was over 100 calls that was handle > good, I was just curios what will happen. Tomorrow I will make real > testes. My production works on 2 core P4 and I have there only > 35 agents CPU load is like 7% with 15% small peeks. > > All phones are sip or analog via sip gateways, PRI is currently > still on asterisk which is connected via sip. > > > /b > > > > On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: > > > >> I did not described it perfectly. I made agents, queues scenarios on > >> conferences. > >> This what I tested was for example 100 calls, so it's 200 channels, > >> and 100 conferences, 2 channels per conference, all are > >> unmuted. I did that just because it is my work scenario. > > > > Brian West > > brian at freeswitch.org > > > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Henry Huang > UniC Solution - Communication Unified > VoIP & Open Source software Consultant > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 elhodred at gmail.com Fri Apr 3 01:19:08 2009 From: elhodred at gmail.com (Alfonso Pinto) Date: Fri, 3 Apr 2009 10:19:08 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls [SOLVED] Message-ID: <8b3b7acc0904030119m264656denaf6b261a398fff27@mail.gmail.com> Hi, Updating asterisk to version 1.4.24 solved the problem. Thanks guys. Regards. 2009/4/2 Brian West : > Follow this > thread?http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html > /b > On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the ?call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=ceKFmNU84B90c. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" ;tag=as26208773. > To: . > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666 at 1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666 at 3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" ;tag=e050QBXFZXN6K. > To: . > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" ;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: . > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666 at 1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" ;tag=as26208773. > To: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: . > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: . > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Contact: . > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia at 1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia at 1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" ;tag=as26208773. > To: ;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666 at 3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" ;tag=e050QBXFZXN6K. > To: ;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > 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 > > Brian West > brian at freeswitch.org > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 elhodred at gmail.com Fri Apr 3 01:22:27 2009 From: elhodred at gmail.com (Alfonso Pinto) Date: Fri, 3 Apr 2009 10:22:27 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls In-Reply-To: <20090402003533.GA9849@jdc.jasonjgw.net> References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com> <64A21B5B-579C-49BA-B869-7164ACADB486@freeswitch.org> <8b3b7acc0904011709q3b8b51c0l27d528243592ccb0@mail.gmail.com> <20090402003533.GA9849@jdc.jasonjgw.net> Message-ID: <8b3b7acc0904030122v4a7ab910j79a6730adea59754@mail.gmail.com> Thank you so much, gmane gives me correct results. Instead, trying to search the thread Brian emailed to me with site:lists.freeswitch.org doesn't give the correct response, thread doesn't appears. Regards 2009/4/2 Jason White : > Alfonso Pinto wrote: >> One question more, maybe a stupid one: How can I search the archives? > > http://www.gmane.org/ > > The searching tool they use, Xapian, tends to give good relevance ranking, at > least in my experience. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 solko at gcdf.pl Fri Apr 3 03:05:30 2009 From: solko at gcdf.pl (Szymon Olko) Date: Fri, 03 Apr 2009 12:05:30 +0200 Subject: [Freeswitch-users] Slow freeswitch shutdown Message-ID: <49D5DF6A.4010204@gcdf.pl> In last SVN trunk version i noticed that stopping of freeswitch takes much time. I have configuration installed with freeswitch. I added sip gateway to my asterisk instance. I don't use asterisk currently and my gateway definition is like that: Starting freeswitch and shutting it down for console with '...' brings following logs. 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() UN-Registering example.com 2009-04-03 11:58:03 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting for worker thread 2009-04-03 11:58:03 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() deleted gateway example.com 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() UN-Registering 429956 2009-04-03 11:58:33 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting for worker thread 2009-04-03 11:58:33 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() deleted gateway 429956 Asterisk was not run at all so it should not register to it, why it hangs to unregister it? From codecomplete at free.fr Fri Apr 3 03:07:36 2009 From: codecomplete at free.fr (Fred) Date: Fri, 03 Apr 2009 12:07:36 +0200 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt Message-ID: <7.0.1.0.2.20090403120452.024273d8@fredshack.com> Carlos Talbot > Is there an interest in running FreeSWITCH on OpenWRT? I recently managed to compile and run a version for a MIPs based router, the Planex MZK-W04NU. Great news :-) I'm interested in running FS on any of this type of small hardware. Ideally, it should have a USB port so I can connect Sangoma's U100 connector to handle one or two POTS lines. Would the FS port you did handle this USB VoIP gateway? Thanks. From andy at fabulous4.co.uk Fri Apr 3 03:49:14 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Fri, 3 Apr 2009 11:49:14 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: Message-ID: <844E4DA20AAD4AB3B123D3A0572CCB5C@wsandy> Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:40 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please try SVN trunk. /b On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote: Hi Brian, 1.03 Thanks Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/c6ca6d2e/attachment-0002.html From yivzhenko at mksat.net Fri Apr 3 05:43:51 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko (WP)) Date: Fri, 3 Apr 2009 15:43:51 +0300 Subject: [Freeswitch-users] Use rates from lcr in nibblebill module In-Reply-To: References: <200904021322.05690.yivzhenko@mksat.net> Message-ID: <200904031543.52605.yivzhenko@mksat.net> Thanks for variables and explanation. Work fine! Now wait for nibblebill can hangup connection when balance hits 0.00 On Thursday 02 April 2009 15:37:28 Rupa Schomaker wrote: > Update the to the latest. I've added more channel vars: > > eg: > > after doing: > > > (not a real number) > > I get the following: > > variable_lcr_query_digits: [12148267722] > variable_lcr_query_profile: [0] > variable_lcr_query_expanded_digits: [12148267722, 1214826772, 121482677, > 12148267, 1214826, 121482, 12148, 1214, 121, 12, 1] > variable_lcr_route_1: > [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gate >way/teliax/12148267722] variable_lcr_rate_1: [0.01000] > variable_lcr_carrier_1: [teliax] > variable_lcr_codec_1: [PCMU] > variable_lcr_route_2: > [[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec_string=PCMU]sofia/ga >teway/vitelity/12148267722] variable_lcr_rate_2: [0.01440] > variable_lcr_carrier_2: [vitelity] > variable_lcr_codec_2: [PCMU] > variable_lcr_route_count: [2] > variable_lcr_auto_route: > [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gate >way/teliax/12148267722|[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec >_string=PCMU]sofia/gateway/vitelity/12148267722] variable_import: > [lcr_carrier,lcr_rate] > > which, I think is what you are asking for. If you know which route you are > going to use (eg: 1) then you can get it's rate by using lcr_rate_1. > > Alternatively, you can use the lcr_auto_route and then once the b-leg > connects, query the b-leg variable for lcr_carrier and lcr_rate to see > which one was actually used. > > You really can't use lcr_auto_route and set a single rate since each leg > can be rated differently (look at example above). > > Normally lcr is used for your own rates between you and your carrier. That > is independant of the rate table used for your customers. You can use lcr > to query both. First use lcr to query your own rates using a different > profile. This would return a *single* route if you setup your route table > right. Save the rate in a var to be used with nibble bill. Then use lcr > with your external rates so you can route according to your own cost with > your carrier(s). > > On Thu, Apr 2, 2009 at 5:22 AM, Yuriy Ivzhenko (WP) wrote: > > Hi, > > > > I want to use module lcr to find a best route and his rate , then make a > > call > > and bill on that rate with nibblebill module. > > > > lcr return variable "lcr_auto_route" that contains "[lcr_rate=xxx]" > > variable > > for new channel. > > To use nibblebill i need to set "nibble_rate" = "lcr_rate". > > > > What is best method to do that? > > Is there a way to do that with standard tools, without use external > > scripts? > > > > > > Thanks, > > Yuriy > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 Apr 3 06:14:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 3 Apr 2009 08:14:41 -0500 Subject: [Freeswitch-users] Slow freeswitch shutdown In-Reply-To: <49D5DF6A.4010204@gcdf.pl> References: <49D5DF6A.4010204@gcdf.pl> Message-ID: <191c3a030904030614i7222eac2k9187c24d5d3e20e3@mail.gmail.com> update again and see if it's better On Fri, Apr 3, 2009 at 5:05 AM, Szymon Olko wrote: > In last SVN trunk version i noticed that stopping of freeswitch takes much > time. > > I have configuration installed with freeswitch. I added sip gateway to my > asterisk instance. I don't use asterisk currently and my > gateway definition is like that: > > > > > > > > > Starting freeswitch and shutting it down for console with '...' brings > following logs. > > 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() > UN-Registering example.com > 2009-04-03 11:58:03 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting > for worker thread > 2009-04-03 11:58:03 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() > deleted gateway example.com > 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() > UN-Registering 429956 > 2009-04-03 11:58:33 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting > for worker thread > 2009-04-03 11:58:33 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() > deleted gateway 429956 > > Asterisk was not run at all so it should not register to it, why it hangs > to unregister 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/30e13bde/attachment-0002.html From lele at windmill.it Fri Apr 3 06:20:13 2009 From: lele at windmill.it (Lele Forzani) Date: Fri, 03 Apr 2009 15:20:13 +0200 Subject: [Freeswitch-users] codecs initialization flags in endpoint modules Message-ID: <1238764813.23024.102.camel@rivendell.windmill.it> Hello, I've been experimenting with the use of mod_dahdi_codec and other ways to perform external transcoding for codecs, and came up with noticing that transcoding resources seemed to be used up twice what I expected. That is and 2x the number of call legs, ending up to two encoder and two decoder instances per leg. So, I looked at the code and noticed almost every endpoint module does something like this (excerpt from mod_sofia, sofia_glue.c:~1800): if (switch_core_codec_init(&tech_pvt->read_codec, tech_pvt->iananame, tech_pvt->rm_fmtp, tech_pvt->rm_rate, tech_pvt->codec_ms, 1, SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | tech_pvt->profile->codec_flags, NULL, switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load codec?\n"); switch_goto_status(SWITCH_STATUS_FALSE, end); } if (switch_core_codec_init(&tech_pvt->write_codec, tech_pvt->iananame, tech_pvt->rm_fmtp, tech_pvt->rm_rate, tech_pvt->codec_ms, 1, SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | tech_pvt->profile->codec_flags, NULL, switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load codec?\n"); switch_goto_status(SWITCH_STATUS_FALSE, end); } The flags being SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE seems to be causing the apparent 'double' allocation of transcoding resources, and I fail to understand the need for both, in both cases. Could someone please spend a minute to explain? thanks lele From pablosaro at gmail.com Fri Apr 3 06:39:45 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 3 Apr 2009 10:39:45 -0300 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> Message-ID: <247f8100904030639l1f076cdt2f0f53303a236cc8@mail.gmail.com> Hi Ashley, A very simple HA solution can be achieved by using SRV. But according to your email, the solution that comes to my mind is the following: PSTN Gw --> OpenSIPs stateless w/ dispatcher module --> many FS boxes And if you want a balanced distribution of the calls, you can write a piece of code to keep statistics of your active sessions in a db. Each time a call arrives to a FS box, you trigger your piece of code to store a session record in a db and when the call ends you update the statistics in the db. This way, OpenSIPs can ask this db before making the decision where to route an incoming call. Fail over? If OpenSIPs gets a time out, just try with the next FS box. I hope it helps you. Pablo 2009/4/2 Ashley van Gerven > Hi, > > I can't find much info on setting up a redundant or heavy load FreeSwitch > implementation. Are there any > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? > > I imagine the entry level solution is to have two FS boxes configured > identitcally, with > redundant SBC software (recommendations?) in front, passing the calls to > the primary FS box, > or the backup FS box if the primary is not responding. Is that the easiest > solution? > > What about a situation of having a level of concurrent calls beyond what > one FS box can handle? I realise > that would be a very large number of concurrent calls, but we would need a > good plan on how to scale the > systems. > > Are there recommendations for load balancing solutions? Either soft or > hardware? > > My guess would be having 3 + 1 spare FS servers would work, where calls are > distributed accross 3 FS boxes > by a load balancer with one spare in event of failure. > > Also how would a FS box at max capacity behave? Does FS monitor available > resources and reject the > excess calls that it can't handle? Or would the load balancer have to be > configured with the maximum number > of calls per box? > > Would love to hear some experiences of deploying FS with failover & high > load. > > > Thanks > Ash > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/4bb02c9c/attachment-0002.html From freeswitch-users at digitaldan.com Fri Apr 3 06:51:35 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Fri, 3 Apr 2009 07:51:35 -0600 (MDT) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <2659508.3491238766312290.JavaMail.daniel@radio> Message-ID: <24754670.3511238766668728.JavaMail.daniel@radio> I have my burst rate set to something low, 4096 right now. I also wrote a flash/flex app that has the same size buffer which results in the audio being heard immediately when connecting. As far as the audio being real time, the audio stream is about 6 seconds behind which I'm guessing is the result of the size of the lame buffers in the mod_shout modules (i'm using g.711 ulaw), I was going to look into that next week. Anyone have any thoughts about where else the delay may be happening? I hoping to get this down to around 2 seconds. D- ----- Original Message ----- From: "Rupa Schomaker" To: freeswitch-users at lists.freeswitch.org Sent: Thursday, April 2, 2009 8:05:38 AM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk 2009/4/2 Anthony Minessale < anthony.minessale at gmail.com > Its the buffering and startup of the shout stream taking up the time, HINT put the shoutcast stream into a local stream with a .loc file and then play that in the conference. Ah, that is easy enough! Though I think with icecast doing the burst_on_connect thingie there should be enough data (pushed much faster than real time) to fill FS's buffers. But that would require mod_shout to cooperate with that strategy. ie: on connect, drain the socket as fast as it can filling it's own buffers. Once it's own buffers are full start streaming. I think right now it drains the socket only as fast as it needs to. Or maybe not. I'll go the local stream route for now.... -- -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/20090403/f0de32fa/attachment-0002.html From stormin.normin at hotmail.co.uk Fri Apr 3 07:02:45 2009 From: stormin.normin at hotmail.co.uk (Stromin Normin) Date: Fri, 3 Apr 2009 15:02:45 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: Thanks for all your help, I finally resolved the issue by setting comfort-noise to false in the conference.conf.xml. From: stormin.normin at hotmail.co.uk To: freeswitch-users at lists.freeswitch.org Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. " Upgrade to Internet Explorer 8 Optimised for MSN. " Download Now _________________________________________________________________ Share your photos with Windows Live Photos ? Free. http://clk.atdmt.com/UKM/go/134665338/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/1218688b/attachment-0002.html From brian at freeswitch.org Fri Apr 3 07:11:59 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Apr 2009 09:11:59 -0500 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: Did it sound more like a machine gun? /b On Apr 3, 2009, at 9:02 AM, Stromin Normin wrote: > Thanks for all your help, I finally resolved the issue by setting > comfort-noise to false in the conference.conf.xml. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/ff34b814/attachment-0002.html From dujinfang at gmail.com Fri Apr 3 08:58:41 2009 From: dujinfang at gmail.com (dujinfang) Date: Fri, 3 Apr 2009 23:58:41 +0800 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? Message-ID: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> Hi, I have outbound gateways returns 403 or 503 constantly. So I tried to dial out using sofia/gateways/gw1/xxxx|sofia/gateways/gw2/xxxx|sofia/gateways/gw3... for fail over. To make it work, I need to set ignore_early_media=true. However, the caller do need to hear the early media to figure out what's going on. If I set ignore_early_media=false, only the first one tried. A little more detail: The gateway is first tier, if it cannot initiate a PSTN channel returns 403/503 immediately. If it can find a PSTN channel, but the callee fails, no answer or busy or others, it plays early media and returns 503. If I want failover, and the early media, how to do that? Thanks. regards, Seven. From msc at freeswitch.org Fri Apr 3 09:28:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 Apr 2009 09:28:59 -0700 Subject: [Freeswitch-users] codecs initialization flags in endpoint modules In-Reply-To: <1238764813.23024.102.camel@rivendell.windmill.it> References: <1238764813.23024.102.camel@rivendell.windmill.it> Message-ID: <87f2f3b90904030928t46fd697auacdc7d5ad01945a7@mail.gmail.com> FYI, these are good questions but they probably belong on the dev list since they are so technical in nature. :) -MC On Fri, Apr 3, 2009 at 6:20 AM, Lele Forzani wrote: > > Hello, > I've been experimenting with the use of mod_dahdi_codec and other ways > to perform external transcoding for codecs, and came up with noticing > that transcoding resources seemed to be used up twice what I expected. > That is and 2x the number of call legs, ending up to two encoder and two > decoder instances per leg. > > > So, I looked at the code and noticed almost every endpoint module does > something like this (excerpt from mod_sofia, sofia_glue.c:~1800): > > if (switch_core_codec_init(&tech_pvt->read_codec, > tech_pvt->iananame, > tech_pvt->rm_fmtp, > tech_pvt->rm_rate, > tech_pvt->codec_ms, > 1, > SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | > tech_pvt->profile->codec_flags, > NULL, switch_core_session_get_pool(tech_pvt->session)) != > SWITCH_STATUS_SUCCESS) { > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load > codec?\n"); > switch_goto_status(SWITCH_STATUS_FALSE, end); > } > > if (switch_core_codec_init(&tech_pvt->write_codec, > tech_pvt->iananame, > tech_pvt->rm_fmtp, > tech_pvt->rm_rate, > tech_pvt->codec_ms, > 1, > SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | > tech_pvt->profile->codec_flags, > NULL, switch_core_session_get_pool(tech_pvt->session)) != > SWITCH_STATUS_SUCCESS) { > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load > codec?\n"); > switch_goto_status(SWITCH_STATUS_FALSE, end); > } > > > The flags being SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE > seems to be causing the apparent 'double' allocation of transcoding > resources, and I fail to understand the need for both, in both cases. > > Could someone please spend a minute to explain? > > > thanks > lele > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/7c1f8511/attachment-0002.html From msc at freeswitch.org Fri Apr 3 09:30:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 Apr 2009 09:30:24 -0700 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: References: Message-ID: <87f2f3b90904030930r2b82a5c3oa9c558b4c5f7052e@mail.gmail.com> On Fri, Apr 3, 2009 at 7:11 AM, Brian West wrote: > Did it sound more like a machine gun? > /b > > Comfort noise for General Douglas McArthur I guess... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/ad368972/attachment-0002.html From brian at freeswitch.org Fri Apr 3 10:04:57 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Apr 2009 12:04:57 -0500 Subject: [Freeswitch-users] MeucciSolutions@66.96.218.5 Message-ID: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> Does anyone else seem to be getting tons of calls from this evil IP? They keep ringing me via SIP over and over again. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/e692a43d/attachment-0002.html From chris.chen2004 at gmail.com Fri Apr 3 10:36:18 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 3 Apr 2009 13:36:18 -0400 Subject: [Freeswitch-users] MeucciSolutions@66.96.218.5 In-Reply-To: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> References: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> Message-ID: <507898380904031036h546a2dc0x39d5927aac431830@mail.gmail.com> Hi Brian, looks like this Evil is calling everywhere today on port 5060, please see my asterisk log [Apr 3 11:13:42] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as05dbf888 [Apr 3 11:25:12] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as5ab1ec7b [Apr 3 11:25:44] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as5ab1ec7b [Apr 3 11:36:17] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as5c4625af [Apr 3 11:55:22] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as4d32ad06 [Apr 3 11:55:54] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as4d32ad06 [Apr 3 11:55:56] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as324c491b [Apr 3 12:00:19] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as4ab90c05 [Apr 3 12:14:43] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as3edfecbb [Apr 3 12:23:38] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as305dbb2e [Apr 3 12:32:14] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as5bf0ab42 [Apr 3 12:49:12] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as7f56ad67 [Apr 3 12:52:21] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as0d5d32e0 [Apr 3 13:10:09] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as1b806860 [Apr 3 13:17:46] NOTICE[16617] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as487f8ecb [Apr 3 13:29:56] NOTICE[16920] chan_sip.c: Failed to authenticate user "MeucciSolutions" >;tag=as613a9814 On Fri, Apr 3, 2009 at 1:04 PM, Brian West wrote: > Does anyone else seem to be getting tons of calls from this evil IP? They > keep ringing me via SIP over and over again. > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/2608f6d1/attachment-0002.html From gkuri at ieee.org Fri Apr 3 10:53:53 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 03 Apr 2009 10:53:53 -0700 Subject: [Freeswitch-users] MeucciSolutions@66.96.218.5 In-Reply-To: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> References: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> Message-ID: <49D64D31.2060904@ieee.org> I heard about this a few days ago, they claim it's not them, but someone trying to "harm their reputation" ... http://www.meucci-solutions.com/complaints.asp?id=1 Gabe Brian West wrote: > Does anyone else seem to be getting tons of calls from this evil IP? > They keep ringing me via SIP over and over again. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 chris.chen2004 at gmail.com Fri Apr 3 11:02:15 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 3 Apr 2009 14:02:15 -0400 Subject: [Freeswitch-users] MeucciSolutions@66.96.218.5 In-Reply-To: <49D64D31.2060904@ieee.org> References: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> <49D64D31.2060904@ieee.org> Message-ID: <507898380904031102ieb3d6e4p9228fa728a5b2e1@mail.gmail.com> It is strange this IP is from US 66.96.218.5USUNITED STATESPENNSYLVANIASCRANTONNETWORK OPERATIONS CENTER INC On Fri, Apr 3, 2009 at 1:53 PM, Gabriel Kuri wrote: > I heard about this a few days ago, they claim it's not them, but someone > trying to "harm their reputation" ... > > http://www.meucci-solutions.com/complaints.asp?id=1 > > Gabe > > Brian West wrote: > > Does anyone else seem to be getting tons of calls from this evil IP? > > They keep ringing me via SIP over and over again. > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/86670c84/attachment-0002.html From brian at freeswitch.org Fri Apr 3 11:09:55 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Apr 2009 13:09:55 -0500 Subject: [Freeswitch-users] MeucciSolutions@66.96.218.5 In-Reply-To: <507898380904031102ieb3d6e4p9228fa728a5b2e1@mail.gmail.com> References: <0957E2DD-030F-40AF-9E4E-CF155AF739B5@freeswitch.org> <49D64D31.2060904@ieee.org> <507898380904031102ieb3d6e4p9228fa728a5b2e1@mail.gmail.com> Message-ID: <9F8ABE86-EE2A-4671-BFEE-E60A78047D76@freeswitch.org> Yes I opened a ticket with them about it... they said it would take 24 hours to figure anything out! /b On Apr 3, 2009, at 1:02 PM, Chris Chen wrote: > It is strange this IP is from US > 66.96.218.5 US UNITED STATES PENNSYLVANIA SCRANTON NETWORK > OPERATIONS CENTER INC > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/dd69580d/attachment-0002.html From carlos.talbot at gmail.com Fri Apr 3 14:29:01 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Fri, 3 Apr 2009 16:29:01 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <7.0.1.0.2.20090403120452.024273d8@fredshack.com> References: <7.0.1.0.2.20090403120452.024273d8@fredshack.com> Message-ID: <5800526b0904031429s3b1deb4do13ecf3335e18949a@mail.gmail.com> This would be ideal. I'm not sure though if the wanpipe kernel driver has been ported to openwrt (or non-x86 hardware for that matter). FYI, I'm slowly working on the wiki and have faced some obstacles as openwrt.org decided to upgrade their servers this past week and have been offline for a good part of that... Carlos On Fri, Apr 3, 2009 at 5:07 AM, Fred wrote: > Carlos Talbot > Is there an interest in running FreeSWITCH on > OpenWRT? I recently managed to compile and run a version for a MIPs > based router, the Planex MZK-W04NU. > > Great news :-) I'm interested in running FS on any of this type of > small hardware. Ideally, it should have a USB port so I can connect > Sangoma's U100 connector to handle one or two POTS lines. > > Would the FS port you did handle this USB VoIP gateway? > > 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/20090403/29d60877/attachment-0002.html From kristian.kielhofner at gmail.com Fri Apr 3 14:42:29 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 3 Apr 2009 17:42:29 -0400 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> Message-ID: <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> You could try (although it's somewhat bleeding edge) to use OpenSIPS 1.5 with load_balancer (not heavily tested, btw) in front of some FreeSWITCH machines: http://www.opensips.org/html/docs/modules/devel/load_balancer.html 2009/4/2 Ashley van Gerven : > Hi, > > I can't find much info on setting up a redundant or heavy load FreeSwitch > implementation. Are there any > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? > > I imagine the entry level solution is to have two FS boxes configured > identitcally, with > redundant SBC software (recommendations?) in front, passing the calls to the > primary FS box, > or the backup FS box if the primary is not responding. Is that the easiest > solution? > > What about a situation of having a level of concurrent calls beyond what one > FS box can handle? I realise > that would be a very large number of concurrent calls, but we would need a > good plan on how to scale the > systems. > > Are there recommendations for load balancing solutions? Either soft or > hardware? > > My guess would be having 3 + 1 spare FS servers would work, where calls are > distributed accross 3 FS boxes > by a load balancer with one spare in event of failure. > > Also how would a FS box at max capacity behave? Does FS monitor available > resources and reject the > excess calls that it can't handle? Or would the load balancer have to be > configured with the maximum number > of calls per box? > > Would love to hear some experiences of deploying FS with failover & high > load. > > > Thanks > Ash > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From pablosaro at gmail.com Fri Apr 3 15:30:24 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 3 Apr 2009 19:30:24 -0300 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> Message-ID: <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> Hi Kristian, you're right. Definitively that will be best solution as soon as it's released as stable (it's alpha now). http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing Pablo On Fri, Apr 3, 2009 at 6:42 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > You could try (although it's somewhat bleeding edge) to use OpenSIPS > 1.5 with load_balancer (not heavily tested, btw) in front of some > FreeSWITCH machines: > > http://www.opensips.org/html/docs/modules/devel/load_balancer.html > > 2009/4/2 Ashley van Gerven : > > Hi, > > > > I can't find much info on setting up a redundant or heavy load FreeSwitch > > implementation. Are there any > > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment? > > > > I imagine the entry level solution is to have two FS boxes configured > > identitcally, with > > redundant SBC software (recommendations?) in front, passing the calls to > the > > primary FS box, > > or the backup FS box if the primary is not responding. Is that the > easiest > > solution? > > > > What about a situation of having a level of concurrent calls beyond what > one > > FS box can handle? I realise > > that would be a very large number of concurrent calls, but we would need > a > > good plan on how to scale the > > systems. > > > > Are there recommendations for load balancing solutions? Either soft or > > hardware? > > > > My guess would be having 3 + 1 spare FS servers would work, where calls > are > > distributed accross 3 FS boxes > > by a load balancer with one spare in event of failure. > > > > Also how would a FS box at max capacity behave? Does FS monitor available > > resources and reject the > > excess calls that it can't handle? Or would the load balancer have to be > > configured with the maximum number > > of calls per box? > > > > Would love to hear some experiences of deploying FS with failover & high > > load. > > > > > > Thanks > > Ash > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/4e87c223/attachment-0002.html From grevenx at me.com Fri Apr 3 15:48:00 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Sat, 04 Apr 2009 00:48:00 +0200 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> Message-ID: Where do you guys read that it's in alpha? On the opensips.org they proclaim OpenSips 1.5 released, with that module being one of the new features. I don't see any mention of it being alpha/beta functionality? Best regards, Even Andr? On 4. april. 2009, at 00.30, Pablo Hernan Saro wrote: > Hi Kristian, you're right. Definitively that will be best solution > as soon as it's released as stable (it's alpha now). > http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing > > Pablo > > On Fri, Apr 3, 2009 at 6:42 PM, Kristian Kielhofner > wrote: > You could try (although it's somewhat bleeding edge) to use OpenSIPS > 1.5 with load_balancer (not heavily tested, btw) in front of some > FreeSWITCH machines: > > http://www.opensips.org/html/docs/modules/devel/load_balancer.html > > 2009/4/2 Ashley van Gerven : > > Hi, > > > > I can't find much info on setting up a redundant or heavy load > FreeSwitch > > implementation. Are there any > > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment > ? > > > > I imagine the entry level solution is to have two FS boxes > configured > > identitcally, with > > redundant SBC software (recommendations?) in front, passing the > calls to the > > primary FS box, > > or the backup FS box if the primary is not responding. Is that the > easiest > > solution? > > > > What about a situation of having a level of concurrent calls > beyond what one > > FS box can handle? I realise > > that would be a very large number of concurrent calls, but we > would need a > > good plan on how to scale the > > systems. > > > > Are there recommendations for load balancing solutions? Either > soft or > > hardware? > > > > My guess would be having 3 + 1 spare FS servers would work, where > calls are > > distributed accross 3 FS boxes > > by a load balancer with one spare in event of failure. > > > > Also how would a FS box at max capacity behave? Does FS monitor > available > > resources and reject the > > excess calls that it can't handle? Or would the load balancer have > to be > > configured with the maximum number > > of calls per box? > > > > Would love to hear some experiences of deploying FS with failover > & high > > load. > > > > > > Thanks > > Ash > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090404/36d7c009/attachment-0002.html From pablosaro at gmail.com Fri Apr 3 16:24:58 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 3 Apr 2009 20:24:58 -0300 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> Message-ID: <247f8100904031624s4a4ea40v4a5c0fd6edd71b42@mail.gmail.com> Not opensips but the module is in alpha. In the modules doc page says "alpha/new". Pablo On 4/3/09, Even Andr? Fiskvik wrote: > Where do you guys read that it's in alpha? > On the opensips.org they proclaim OpenSips 1.5 released, > with that module being one of the new features. I don't see any > mention of it being alpha/beta functionality? > > Best regards, > Even Andr? > > On 4. april. 2009, at 00.30, Pablo Hernan Saro wrote: > >> Hi Kristian, you're right. Definitively that will be best solution >> as soon as it's released as stable (it's alpha now). >> http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing >> >> Pablo >> >> On Fri, Apr 3, 2009 at 6:42 PM, Kristian Kielhofner >> > > wrote: >> You could try (although it's somewhat bleeding edge) to use OpenSIPS >> 1.5 with load_balancer (not heavily tested, btw) in front of some >> FreeSWITCH machines: >> >> http://www.opensips.org/html/docs/modules/devel/load_balancer.html >> >> 2009/4/2 Ashley van Gerven : >> > Hi, >> > >> > I can't find much info on setting up a redundant or heavy load >> FreeSwitch >> > implementation. Are there any >> > links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment >> ? >> > >> > I imagine the entry level solution is to have two FS boxes >> configured >> > identitcally, with >> > redundant SBC software (recommendations?) in front, passing the >> calls to the >> > primary FS box, >> > or the backup FS box if the primary is not responding. Is that the >> easiest >> > solution? >> > >> > What about a situation of having a level of concurrent calls >> beyond what one >> > FS box can handle? I realise >> > that would be a very large number of concurrent calls, but we >> would need a >> > good plan on how to scale the >> > systems. >> > >> > Are there recommendations for load balancing solutions? Either >> soft or >> > hardware? >> > >> > My guess would be having 3 + 1 spare FS servers would work, where >> calls are >> > distributed accross 3 FS boxes >> > by a load balancer with one spare in event of failure. >> > >> > Also how would a FS box at max capacity behave? Does FS monitor >> available >> > resources and reject the >> > excess calls that it can't handle? Or would the load balancer have >> to be >> > configured with the maximum number >> > of calls per box? >> > >> > Would love to hear some experiences of deploying FS with failover >> & high >> > load. >> > >> > >> > Thanks >> > Ash >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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://blog.krisk.org >> http://www.submityoursip.com >> http://www.astlinux.org >> http://www.star2star.com >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 Gmail for mobile | mobile.google.com From jason at jasonjgw.net Fri Apr 3 16:53:00 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 4 Apr 2009 10:53:00 +1100 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> Message-ID: <20090403235300.GA10045@jdc.jasonjgw.net> dujinfang wrote: > However, the caller do need to hear the early media to figure out > what's going on. If I set ignore_early_media=false, only the first one > tried. Could you use ring_ready? that way, the calling SIP phone should generate the ringback. From brian at freeswitch.org Fri Apr 3 17:13:57 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Apr 2009 19:13:57 -0500 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <20090403235300.GA10045@jdc.jasonjgw.net> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> Message-ID: <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> First one to give media wins unless you ignore_early_media /b On Apr 3, 2009, at 6:53 PM, Jason White wrote: > Could you use ring_ready? that way, the calling SIP phone should > generate the > ringback. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090403/e7881c46/attachment-0002.html From kristian.kielhofner at gmail.com Fri Apr 3 17:45:27 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 3 Apr 2009 20:45:27 -0400 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> Message-ID: <2d9149cd0904031745r4f5bc194hc7718159c1875b0d@mail.gmail.com> Pablo, It is very cool and a very compelling reason to upgrade/move to OpenSIPS 1.5. I'm running (mostly) OpenSIPS 1.4.4/1.4.5 now and it's rock solid (as usual). It's really an excellent complement to FreeSWITCH! I will be doing testing with 1.5 and the new load balancer module shortly. On Fri, Apr 3, 2009 at 6:30 PM, Pablo Hernan Saro wrote: > Hi Kristian, you're right. Definitively that will be best solution as soon > as it's released as stable (it's alpha now). > http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing > > Pablo > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From pablosaro at gmail.com Fri Apr 3 20:48:06 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Sat, 4 Apr 2009 00:48:06 -0300 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <2d9149cd0904031745r4f5bc194hc7718159c1875b0d@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> <2d9149cd0904031745r4f5bc194hc7718159c1875b0d@mail.gmail.com> Message-ID: <247f8100904032048v316454b5rceb685b816d88477@mail.gmail.com> Hi Kristian, Let us know your experience as soon as you try it. Why not write a wiki page? =) On Fri, Apr 3, 2009 at 9:45 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Pablo, > > It is very cool and a very compelling reason to upgrade/move to > OpenSIPS 1.5. I'm running (mostly) OpenSIPS 1.4.4/1.4.5 now and it's > rock solid (as usual). It's really an excellent complement to > FreeSWITCH! > > I will be doing testing with 1.5 and the new load balancer module shortly. > > On Fri, Apr 3, 2009 at 6:30 PM, Pablo Hernan Saro > wrote: > > Hi Kristian, you're right. Definitively that will be best solution as > soon > > as it's released as stable (it's alpha now). > > http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing > > > > Pablo > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090404/42375ca4/attachment-0002.html From zhaoxxqq at 163.com Fri Apr 3 22:33:07 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Sat, 4 Apr 2009 13:33:07 +0800 Subject: [Freeswitch-users] compile problem in vista. Message-ID: <200904041333057523168@163.com> Hi, It's first time I install FS in Vista. After having downloaded the FS sources from svn. Follow the instruction on how to build FS on Windows. I Using Visual C++ 2008 Express Open Freeswitch.sln Right click the main solution node at the top of the Solution Explorer Right click and select Build after do this I was stoped by the problem. the error is like below, what need I to do? anyone can help me? Error 6 error C2008: '#' : unexpected in macro definition c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.h 1532 Error 8 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 9 error C2065: 'defiTE_a_15' : undeclared identifier c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 10 error C2099: initializer is not a constant c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 11 error C2061: syntax error : identifier 'defiTE_a_15' c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 12 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 13 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 14 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 15 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 16 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 17 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 18 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 19 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 20 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 21 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 22 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 23 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 24 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal.c': No such file or directory c1 Error 25 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_diphone.c': No such file or directory c1 Error 26 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_lpc.c': No such file or directory c1 Error 27 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_res.c': No such file or directory c1 Error 28 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_residx.c': No such file or directory c1 Error 30 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_aswd.c': No such file or directory c1 Error 31 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_dur_stats.c': No such file or directory c1 Error 32 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_durz_cart.c': No such file or directory c1 Error 33 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_expand.c': No such file or directory c1 Error 34 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_f0_model.c': No such file or directory c1 Error 35 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_f0lr.c': No such file or directory c1 Error 36 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_ffeatures.c': No such file or directory c1 Error 37 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_gpos.c': No such file or directory c1 Error 38 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_int_accent_cart.c': No such file or directory c1 Error 39 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_int_tone_cart.c': No such file or directory c1 Error 40 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_nums_cart.c': No such file or directory c1 Error 41 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_phoneset.c': No such file or directory c1 Error 42 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_phrasing_cart.c': No such file or directory c1 Error 43 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_text.c': No such file or directory c1 Error 44 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\usenglish.c': No such file or directory c1 Error 47 fatal error C1083: Cannot open include file: 'usenglish.h': No such file or directory c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt.c 46 Error 103 fatal error C1083: Cannot open include file: 'usenglish.h': No such file or directory c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb.c 46 Error 109 fatal error C1083: Cannot open include file: 'usenglish.h': No such file or directory c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms.c 46 Error 123 error C2220: warning treated as error - no 'object' file generated c:\Users\lenovo\Documents\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua.c 1 Error 140 error C2695: 'LUA::Session::destroy': overriding virtual function differs from 'CoreSession::destroy' only by calling convention c:\users\lenovo\documents\freeswitch\src\mod\languages\mod_lua\freeswitch_lua.h 26 Error 149 fatal error LNK1181: cannot open input file 'flite.lib' mod_flite Error 175 fatal error LNK1181: cannot open input file '..\..\..\..\libs\win32\sofia\debug\libsofia_sip_ua_static.lib' mod_sofia 2009-04-04 zhaoxxqq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090404/87880a4e/attachment-0002.html From dujinfang at gmail.com Fri Apr 3 23:35:18 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 4 Apr 2009 14:35:18 +0800 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <20090403235300.GA10045@jdc.jasonjgw.net> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> Message-ID: On Apr 4, 2009, at 7:53 AM, Jason White wrote: > dujinfang wrote: >> However, the caller do need to hear the early media to figure out >> what's going on. If I set ignore_early_media=false, only the first >> one >> tried. > > Could you use ring_ready? that way, the calling SIP phone should > generate the > ringback. > ring_ready would be replaced by remote party early media. It does not harm though, I still need to listen early media. 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 dujinfang at gmail.com Fri Apr 3 23:41:03 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 4 Apr 2009 14:41:03 +0800 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> Message-ID: On Apr 4, 2009, at 8:13 AM, Brian West wrote: > First one to give media wins unless you ignore_early_media > > /b > Thanks, I tested again. That's exactly what I want except the problem sometimes the gateway gives (wrong)early_media but fails immediately, so I have no chance to hear the early media. And unfortunately the gateway is beyond my control. :( Will do more testing. > > On Apr 3, 2009, at 6:53 PM, Jason White wrote: > >> Could you use ring_ready? that way, the calling SIP phone should >> generate the >> ringback. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090404/cec0e888/attachment-0002.html From kristian.kielhofner at gmail.com Sat Apr 4 00:13:44 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Sat, 4 Apr 2009 03:13:44 -0400 Subject: [Freeswitch-users] FS failover redundancy & load balancing In-Reply-To: <247f8100904032048v316454b5rceb685b816d88477@mail.gmail.com> References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com> <2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com> <247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com> <2d9149cd0904031745r4f5bc194hc7718159c1875b0d@mail.gmail.com> <247f8100904032048v316454b5rceb685b816d88477@mail.gmail.com> Message-ID: <2d9149cd0904040013u180eed38q4c1ff09dd8587487@mail.gmail.com> Hey Pablo, Wiki page? I just might! :) On Fri, Apr 3, 2009 at 11:48 PM, Pablo Hernan Saro wrote: > Hi Kristian, > > Let us know your experience as soon as you try it. Why not write a wiki > page?? =) > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Sat Apr 4 00:32:38 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Sat, 4 Apr 2009 03:32:38 -0400 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> Message-ID: <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> On Sat, Apr 4, 2009 at 2:41 AM, dujinfang wrote: > > On Apr 4, 2009, at 8:13 AM, Brian West wrote: > > First one to give media wins unless you ignore_early_media > /b > > Thanks, I tested again. That's exactly what I want except the problem > sometimes the gateway gives (wrong)early_media but fails immediately, so I > have no chance to hear the early media. And unfortunately the gateway is > beyond my control. :( > Will do more testing. > I'm not really sure how else FS should handle this... As Brian said "first one with media wins". That's the problem with early media. Is it ringback that might turn into a completed call? Is it some error message played to the user? Is it someones voicemail system, trying to save some money? One way or another, is it something the user should hear? No way to know, really... 180/183 with SDP is a bit ambiguous. I always get frustrated when various people /insist/ on using 183 w/ SDP just for ringback. Have they never heard of 180 w/o SDP? Let me generate my own local ringback and/or handle the situation accordingly! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From dave at 3c.co.uk Sun Apr 5 20:12:17 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 06 Apr 2009 04:12:17 +0100 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> Message-ID: <1238987537.13971.13.camel@dk-d820> On Sat, 2009-04-04 at 03:32 -0400, Kristian Kielhofner wrote: > > 180/183 with SDP is a bit ambiguous. I always get frustrated when > various people /insist/ on using 183 w/ SDP just for ringback. Have > they never heard of 180 w/o SDP? Let me generate my own local > ringback and/or handle the situation accordingly! Ah, well, that's where you're trying to change the way that things have been done for some decades. Ringback has historically been generated close to the called party, which is why you hear different ringback if you call people in different countries. Furthermore, that audio path is used to convey all sorts of messages: "the number you have called has been changed", "the cellphone you have called has not responded", "calls to 1-800 numbers are not free if made from overseas.." Lastly, there's no guarantee that it'll be possible to differentiate between one of these and ringback from the signalling alone and, in many cases, there is simply no mechanism available to provide such differentiation. You're probably best advised to swim with the tide on this one..! Cheers -- Dave From brian at freeswitch.org Sun Apr 5 20:29:34 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 5 Apr 2009 22:29:34 -0500 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <1238987537.13971.13.camel@dk-d820> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> Message-ID: <2DCE0758-C4F4-4EEB-9808-DBA7344DF48B@freeswitch.org> Yes there was till the SIP RFC writers happen to make their ears rather sore! (RCI) 180 vs 183 should have been it... but NO they had to be ambiguous about that too... if you get a 180 without an SDP you generate... 180 or 183 with SDP (they had a sense of humor about this one I think!) Then this one tops the cake... on early media with forked dial... Say you call billy, mary and ken at the same time. Billy's address provides early media (ringing) you are to give the first one to respond with media to the caller... but if by chance Mary's phone provider is having a problem and they give congestion 20 seconds later and actually answer the call to do this cuz you know how stupid telco's are... now you are to give the caller the congestion tone... So you had prefect ringing.. then congestion... I think we have all be there, heard that! /b On Apr 5, 2009, at 10:12 PM, David Knell wrote: > signalling alone and, in many cases, there is simply no mechanism > available to provide such differentiation. From kristian.kielhofner at gmail.com Sun Apr 5 21:08:16 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 6 Apr 2009 00:08:16 -0400 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <1238987537.13971.13.camel@dk-d820> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> Message-ID: <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> On Sun, Apr 5, 2009 at 11:12 PM, David Knell wrote: > > Ah, well, that's where you're trying to change the way that things > have been done for some decades. ?Ringback has historically been > generated close to the called party, which is why you hear different > ringback if you call people in different countries. What's wrong with that? Isn't that what we are all doing (or trying to do), to some extent? International dialing very well may use different ringbacks but: 1) How important is this, really? 2) How much more complicated is adding at least the real potential for 180? Actually using 180 w/o SDP provides for enhanced call handing functionality while only requiring (in many cases) one additional test scenario. Consider the current example (all 180s are actually 180s w/o SDP and 183 is 183 w/ SDP): Bridging a call to multiple destinations (A, B, and C). A: 100,180 B: 100,180,200 C: 100,183 We could have implemented proper forking if it weren't for C who insisted on sending media early (for whatever reason). While I could see many scenarios where this might happen even with the configuration I suggest, consider what would happen in the ideal scenario: A: 100,180 B: 100,180,200 C: 100,180 Ah, B won because it was the first endpoint to actually /answer/ the call and begin playing media. Nice and clean. This is what happens when dialing local phones behind a PBX. All endpoint SIP phones send 180 to allow for clean parallel forking across them. This is what makes configuration for ring groups, etc, possible. I'm not suggesting that this configuration could be simply "dropped in" when dialing to the PSTN but it should at least be a a possibility. I suppose the other thing here (which is possible and has been suggested) is to configure your device to ignore early media. Too bad (due to various reasons, some of them being legacy PSTN) that in some cases the user should hear that 183. Speaking of which... > Furthermore, that audio path is used to convey all sorts of messages: > "the number you have called has been changed", "the cellphone you have > called has not responded", "calls to 1-800 numbers are not free if > made from overseas.." ?Lastly, there's no guarantee that it'll be > possible to differentiate between one of these and ringback from the > signalling alone and, in many cases, there is simply no mechanism > available to provide such differentiation. People poke at SIP all the time for this one but this is where the PSTN even seems a bit ambiguous. We have ISDN cause codes AND inband audio messages? I'm reminded of a situation the other day with a provider's SIP architecture. If you send a call to a completely bogus destination number (1, in this case) they reply with an inband audio error message. Why not send a 404 or something that is easily parsed and understood by my platform (FreeSWITCH)? In this case I needed to do some further action in the event of a "call failure" and as far as bridge/mod_sofia is concerned this was a "successful" call. I know this specific instance could be avoided but I can't wait to see what else they play inband audio messages for. Of course I can't really configure my end to ignore early media because I could miss out on some legit inband audio messaging that is actually useful. > You're probably best advised to swim with the tide on this one..! If I "swam with the tide" I'd probably be out getting my CCIE and installing Call Manager systems or something ;). Maybe that's not the best or the most "fair" analogy but hopefully you can see my point. I think there's a little rebel in all of us here on freeswitch-users! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From andy at fabulous4.co.uk Mon Apr 6 02:07:25 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 6 Apr 2009 10:07:25 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: Message-ID: Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:40 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please try SVN trunk. /b On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote: Hi Brian, 1.03 Thanks Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/dfc8dedd/attachment-0002.html From helmut.kuper at ewetel.de Mon Apr 6 03:20:25 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 06 Apr 2009 12:20:25 +0200 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> <49CB9E0C.4030300@ewetel.de> <49CCEC15.8010500@ewetel.de> Message-ID: <49D9D769.5020101@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I still have this problem. From the day of starting up freeswitch two threads are consuming slowly more and more CPU power. In parallel FS virtual and physical memory usage grows slowly as well. FS is up for 6 days now and served 3160 sessions. Virtual memory usage has grown from 200MB to 1.1GB (18,1%) and is still growing. CPU is mostly around 25% with lowest of 17% and a maximum of 50% (all on a 32 bit 4 CPU core system) and is still growing. There are two FS-Threads with nearly same CPU usage of currently around 20% each (I used htop for this): strace Thread 1 (I guess this is the sofia/sip thread): epoll_wait(21, {}, 4, 0) = 0 epoll_wait(21, {}, 4, 0) = 0 epoll_wait(21, {{EPOLLIN, {u32=2, u64=2}}}, 4, 0) = 1 ioctl(24, FIONREAD, [268]) = 0 recvmsg(24, {msg_name(16)={sa_family=AF_INET, sin_port=htons(1068), sin_addr=inet_addr("85.16.245.249")}, msg_iov(1)=[{"SIP/2.0 200 Ok\r\nVia: SIP/2.0/UDP"..., 268}], msg_controllen=0, msg_flags=0}, 0) = 268 gettimeofday({1239012809, 343580}, NULL) = 0 gettimeofday({1239012809, 343645}, NULL) = 0 epoll_wait(21, {}, 4, 0) = 0 epoll_wait(21, {}, 4, 0) = 0 ... strace Thread 2: select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) select(0, NULL, NULL, NULL, {1, 0}) = 0 (Timeout) ... I use FreeSWITCH Version 1.0.trunk (12347M) regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFJ2ddp4tZeNddg3dwRAlkXAJ9fIwpJw6u18JPhFC4hzB+0Z1iAbgCfW7AE dnrmpXDLVOnWtjwFKMoVw48= =zzZ9 -----END PGP SIGNATURE----- -------------- next part -------------- A non-text attachment was scrubbed... Name: 0xD760DDDC.asc Type: application/pgp-keys Size: 1854 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/f512e936/attachment-0004.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: 0xD760DDDC.asc Type: application/pgp-keys Size: 1854 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/f512e936/attachment-0005.bin From steveu at coppice.org Mon Apr 6 04:40:41 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 06 Apr 2009 19:40:41 +0800 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <2DCE0758-C4F4-4EEB-9808-DBA7344DF48B@freeswitch.org> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> <2DCE0758-C4F4-4EEB-9808-DBA7344DF48B@freeswitch.org> Message-ID: <49D9EA39.4010404@coppice.org> Brian West wrote: > Say you call billy, mary and ken at the same time. Billy's address > provides early media (ringing) you are to give the first one to > respond with media to the caller... but if by chance Mary's phone > provider is having a problem and they give congestion 20 seconds later > and actually answer the call to do this cuz you know how stupid > telco's are... now you are to give the caller the congestion tone... > So you had prefect ringing.. then congestion... I think we have all be > there, heard that! > Er, that's not stupidity. If the regulations allow them to answer at this point, they will. It generates revenue. Its a disaster for a lot of services which need to know if the call was answered to tell what to do next, but it ain't done through stupidity. We are the stupid suckers who pay. Steve From codecomplete at free.fr Mon Apr 6 04:41:59 2009 From: codecomplete at free.fr (Fred) Date: Mon, 06 Apr 2009 13:41:59 +0200 Subject: [Freeswitch-users] Freeswitch not started OK at boottime? Message-ID: <7.0.1.0.2.20090406133425.05092870@fredshack.com> Hello I'm having a problem connecting to the Freeswitch server running on a Suse server when the it's started at bootime, but OK if I start it manually through the init.d script, so I guess I did something wrong when setting things up. Here's what I did: 1. Downloaded and compiled the latest SVN source 2. cp /usr/src/freeswitch/build/freeswitch.init.suse /etc/init.d/freeswitch 3. chmod 755 /etc/init.d/freeswitch 4. chkconfig freeswitch 345 5. chkconfig -l freeswitch 6. (why needed in addition to chkconfig?) ln -s /etc/init.d/freeswitch /usr/sbin/rcfreeswitch 7. Edit /etc/init.d/freeswitch: FREESWITCH_BIN=/usr/local/freeswitch/bin/freeswitch #(BAD!) FREESWITCH_CONFIG=/usr/local/freeswitch/conf/freeswitch.xml FREESWITCH_PARAMS="-nc" Here's what it says when I try to connect to the server: ========= # ps aux | grep free root 3497 0.6 0.7 16912 8212 ? Sl 12:03 0:00 /usr/local/freeswitch/bin/freeswitch -nc # cd /usr/local/freeswitch/bin/ # ./fs_cli [ERROR] libs/esl/fs_cli.c:642 main() Error Connecting [Socket Connection Error] ========= Here's how to solve this issue manually: ========= # /etc/init.d/freeswitch stop Shutting down FreeSWITCH done # /etc/init.d/freeswitch start Starting FreeSWITCH 3867 Backgrounding. done /usr/local/freeswitch/bin # ./fs_cli [logo deleted] +OK log level [7] freeswitch at internal> /exit # ========= Any idea what is wrong? Thank you for any hint. From steveu at coppice.org Mon Apr 6 04:43:05 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 06 Apr 2009 19:43:05 +0800 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> Message-ID: <49D9EAC9.8090804@coppice.org> Kristian Kielhofner wrote: > On Sun, Apr 5, 2009 at 11:12 PM, David Knell wrote: > >> Ah, well, that's where you're trying to change the way that things >> have been done for some decades. Ringback has historically been >> generated close to the called party, which is why you hear different >> ringback if you call people in different countries. >> > > What's wrong with that? Isn't that what we are all doing (or trying > to do), to some extent? > > International dialing very well may use different ringbacks but: > > 1) How important is this, really? > 2) How much more complicated is adding at least the real potential for 180? > The actual ringback tone may not be important, but many other supervisory indications may occur at that point, either as tones or as voice announcements. E.g. call a cellphone that has problems - out of range, out of service, etc - and you will probably get a voice announcement telling you want's up. Steve From brian at freeswitch.org Mon Apr 6 06:24:05 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Apr 2009 08:24:05 -0500 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: References: Message-ID: Please update... rebootstrap.. you caught SVN with the libtool patch which kinda broken a few things linking. /b On Apr 6, 2009, at 4:07 AM, Andy Ayers wrote: > Hi Brian, > > I've upgraded to svn trunk but am now getting errors on load which > are preventing it from working: > > 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_shout.so > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: > ogg_stream_pagein** > 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_spidermonkey.so > **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: > PR_LocalTimeParameters** > > Sorry if this is obvious but what have I done wrong? > > Thanks for your help > Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/a1d4bc81/attachment-0002.html From brian at freeswitch.org Mon Apr 6 06:31:41 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Apr 2009 08:31:41 -0500 Subject: [Freeswitch-users] Freeswitch not started OK at boottime? In-Reply-To: <7.0.1.0.2.20090406133425.05092870@fredshack.com> References: <7.0.1.0.2.20090406133425.05092870@fredshack.com> Message-ID: <0FEBBD98-0B4E-43F9-85D5-D7C66D290029@freeswitch.org> What run level are you starting freeswitch? /b On Apr 6, 2009, at 6:41 AM, Fred wrote: > Hello > > I'm having a problem connecting to the Freeswitch server running on a > Suse server when the it's started at bootime, but OK if I start it > manually through the init.d script, so I guess I did something wrong > when setting things up. > > Here's what I did: > 1. Downloaded and compiled the latest SVN source > 2. cp /usr/src/freeswitch/build/freeswitch.init.suse /etc/init.d/ > freeswitch > 3. chmod 755 /etc/init.d/freeswitch > 4. chkconfig freeswitch 345 > 5. chkconfig -l freeswitch > 6. (why needed in addition to chkconfig?) ln -s > /etc/init.d/freeswitch /usr/sbin/rcfreeswitch > 7. Edit /etc/init.d/freeswitch: > FREESWITCH_BIN=/usr/local/freeswitch/bin/freeswitch > #(BAD!) FREESWITCH_CONFIG=/usr/local/freeswitch/conf/freeswitch.xml > FREESWITCH_PARAMS="-nc" From dujinfang at gmail.com Mon Apr 6 06:46:42 2009 From: dujinfang at gmail.com (dujinfang) Date: Mon, 6 Apr 2009 21:46:42 +0800 Subject: [Freeswitch-users] Freeswitch not started OK at boottime? In-Reply-To: <7.0.1.0.2.20090406133425.05092870@fredshack.com> References: <7.0.1.0.2.20090406133425.05092870@fredshack.com> Message-ID: <16CBB37E-E274-4B14-9EAA-CEE1DC679A6B@gmail.com> > Here's what it says when I try to connect to the server: > ========= > # ps aux | grep free > root 3497 0.6 0.7 16912 8212 ? Sl 12:03 0:00 > /usr/local/freeswitch/bin/freeswitch -nc > It seems started, I never used a suse, however, can you try this? #netstat -an | grep 8021 Maybe FS started before network is ready. Check scripts in /etc/ rc.d/ or any equiv > # cd /usr/local/freeswitch/bin/ > # ./fs_cli > [ERROR] libs/esl/fs_cli.c:642 main() Error Connecting [Socket > Connection Error] > ========= From dave at 3c.co.uk Mon Apr 6 06:47:16 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 06 Apr 2009 14:47:16 +0100 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> Message-ID: <1239025636.12559.13.camel@dk-d820> On Mon, 2009-04-06 at 00:08 -0400, Kristian Kielhofner wrote: > Actually using 180 w/o SDP provides for enhanced call handing > functionality while only requiring (in many cases) one additional test > scenario. Consider the current example (all 180s are actually 180s > w/o SDP and 183 is 183 w/ SDP): > > Bridging a call to multiple destinations (A, B, and C). > > A: 100,180 > B: 100,180,200 > C: 100,183 > > We could have implemented proper forking if it weren't for C who > insisted on sending media early (for whatever reason). While I could > see many scenarios where this might happen even with the configuration > I suggest, consider what would happen in the ideal scenario: > > A: 100,180 > B: 100,180,200 > C: 100,180 > Ah, B won because it was the first endpoint to actually /answer/ the > call and begin playing media. Nice and clean. Hang on - if you want to bridge the call on *answer*, then bridge it on answer, not when one leg starts sending you early media. I've no idea if FS supports this behaviour for its forked dialling, but it's easy to do with a bunch of originates, and uuid_bridge the inbound leg to the first one which answers. > People poke at SIP all the time for this one but this is where the > PSTN even seems a bit ambiguous. We have ISDN cause codes AND inband > audio messages? Yes. A clearing code is used when the call's cleared; inband audio can be used to give the caller more information than a simple clearing code might allow - for example, "The number you are calling has been changed. Please redial on whatever the new number might be." It makes eminent sense - simple, common causes (e.g. user busy) get dealt with as part of the call clearing and it's the responsibility of the originating switch to tell the user; more (indeed arbitrarily) complex ones are dealt with by the far end. --Dave From carthick84 at gmail.com Mon Apr 6 07:04:11 2009 From: carthick84 at gmail.com (B Karthik) Date: Mon, 6 Apr 2009 19:34:11 +0530 Subject: [Freeswitch-users] High CPU load but only few sessions Message-ID: It could be due to registrations. I am currently trying to troubleshoot this problem. I used a sipp scenario to authenticate with fs and register about 2000 different accounts (absolutely no calls made on the test setup). Memory usage increases continuously and does not decrease at all and crosses more than 1 GB in a few hours. On the other hand, there is another fs setup with bypass media turned on and no registrations and is up for almost 45 days without restart and has consumed only about 95 MB of memory and twice as much virtual memory. B Karthik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/75a24d68/attachment-0002.html From brian at freeswitch.org Mon Apr 6 07:14:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Apr 2009 09:14:15 -0500 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: References: Message-ID: <6703B4CB-4097-4349-9427-D5B19C6474E7@freeswitch.org> If you guys are not on rev 12914 then you'll need to update. /b On Apr 6, 2009, at 9:04 AM, B Karthik wrote: > It could be due to registrations. I am currently trying to > troubleshoot this problem. I used a sipp scenario to authenticate > with fs and register about 2000 different accounts (absolutely no > calls made on the test setup). Memory usage increases continuously > and does not decrease at all and crosses more than 1 GB in a few > hours. On the other hand, there is another fs setup with bypass > media turned on and no registrations and is up for almost 45 days > without restart and has consumed only about 95 MB of memory and > twice as much virtual memory. > > B Karthik Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/a2fa2196/attachment-0002.html From helmut.kuper at ewetel.de Mon Apr 6 07:32:05 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 06 Apr 2009 16:32:05 +0200 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: References: Message-ID: <49DA1265.4050907@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, in my scenario I have a reregistration interval of 60 seconds and 32 sip phones connected. So I have a good amount of registrations. Additionally each phone subscribes to itself for MWI and some phone subscribes to others for BLF. Registrar database looks fine. No unused entries there. First I will upgrade to recent svn trunk. If that doesn't help, I will run valgrind on my production system and hope that my machine is strong enough to deliver its service even with valrgind. regards Helmut On 06.04.2009 16:04, B Karthik wrote: > It could be due to registrations. I am currently trying to troubleshoot > this problem. I used a sipp scenario to authenticate with fs and > register about 2000 different accounts (absolutely no calls made on the > test setup). Memory usage increases continuously and does not decrease > at all and crosses more than 1 GB in a few hours. On the other hand, > there is another fs setup with bypass media turned on and no > registrations and is up for almost 45 days without restart and has > consumed only about 95 MB of memory and twice as much virtual memory. > > B Karthik > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFJ2hJl4tZeNddg3dwRAmEUAKCTt1aBPlp1pgs3RHw2AVEuH8ixqgCfWkm8 6vsgh6Ha34/gdg6iDEEEOR0= =2H4m -----END PGP SIGNATURE----- From carthick84 at gmail.com Mon Apr 6 08:02:22 2009 From: carthick84 at gmail.com (B Karthik) Date: Mon, 6 Apr 2009 08:02:22 -0700 (PDT) Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49DA1265.4050907@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> Message-ID: <1239030142502-2593558.post@n2.nabble.com> I updated to the latest revision. No Luck -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, in my scenario I have a reregistration interval of 60 seconds and 32 sip phones connected. So I have a good amount of registrations. Additionally each phone subscribes to itself for MWI and some phone subscribes to others for BLF. Registrar database looks fine. No unused entries there. First I will upgrade to recent svn trunk. If that doesn't help, I will run valgrind on my production system and hope that my machine is strong enough to deliver its service even with valrgind. regards Helmut On 06.04.2009 16:04, B Karthik wrote: > It could be due to registrations. I am currently trying to troubleshoot > this problem. I used a sipp scenario to authenticate with fs and > register about 2000 different accounts (absolutely no calls made on the > test setup). Memory usage increases continuously and does not decrease > at all and crosses more than 1 GB in a few hours. On the other hand, > there is another fs setup with bypass media turned on and no > registrations and is up for almost 45 days without restart and has > consumed only about 95 MB of memory and twice as much virtual memory. > > B Karthik > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFJ2hJl4tZeNddg3dwRAmEUAKCTt1aBPlp1pgs3RHw2AVEuH8ixqgCfWkm8 6vsgh6Ha34/gdg6iDEEEOR0= =2H4m -----END PGP SIGNATURE----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593558.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Mon Apr 6 08:21:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 6 Apr 2009 10:21:36 -0500 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <1239030142502-2593558.post@n2.nabble.com> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> <1239030142502-2593558.post@n2.nabble.com> Message-ID: <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> did you both follow the policy to upgrade? stop fs type make current restart fs if you do not rebuild sofia too (only happens in make current) I just fixed all the problems with these symptoms, 38 million registrations in a 2 day span using 62mb btw, did we not make the policy clear enough about not reporting bugs on the mailing list? On Mon, Apr 6, 2009 at 10:02 AM, B Karthik wrote: > > I updated to the latest revision. No Luck > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > in my scenario I have a reregistration interval of 60 seconds and 32 sip > phones connected. So I have a good amount of registrations. Additionally > each phone subscribes to itself for MWI and some phone subscribes to > others for BLF. > > Registrar database looks fine. No unused entries there. > > First I will upgrade to recent svn trunk. If that doesn't help, I will > run valgrind on my production system and hope that my machine is strong > enough to deliver its service even with valrgind. > > regards > Helmut > > > On 06.04.2009 16:04, B Karthik wrote: > > It could be due to registrations. I am currently trying to troubleshoot > > this problem. I used a sipp scenario to authenticate with fs and > > register about 2000 different accounts (absolutely no calls made on the > > test setup). Memory usage increases continuously and does not decrease > > at all and crosses more than 1 GB in a few hours. On the other hand, > > there is another fs setup with bypass media turned on and no > > registrations and is up for almost 45 days without restart and has > > consumed only about 95 MB of memory and twice as much virtual memory. > > > > B Karthik > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFJ2hJl4tZeNddg3dwRAmEUAKCTt1aBPlp1pgs3RHw2AVEuH8ixqgCfWkm8 > 6vsgh6Ha34/gdg6iDEEEOR0= > =2H4m > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > View this message in context: > http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593558.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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/84fc6cba/attachment-0002.html From carthick84 at gmail.com Mon Apr 6 08:30:41 2009 From: carthick84 at gmail.com (B Karthik) Date: Mon, 6 Apr 2009 08:30:41 -0700 (PDT) Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> <1239030142502-2593558.post@n2.nabble.com> <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> Message-ID: <1239031841283-2593704.post@n2.nabble.com> yes, i did exactly as you mentioned. I will try building again from a fresh checkout. I am sorry about not following the policy, I didn't intend to report it as a bug since i was still unsure that it could be a problem in Freeswitch. did you both follow the policy to upgrade? stop fs type make current restart fs if you do not rebuild sofia too (only happens in make current) I just fixed all the problems with these symptoms, 38 million registrations in a 2 day span using 62mb btw, did we not make the policy clear enough about not reporting bugs on the mailing list? On Mon, Apr 6, 2009 at 10:02 AM, B Karthik wrote: > > I updated to the latest revision. No Luck > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > in my scenario I have a reregistration interval of 60 seconds and 32 sip > phones connected. So I have a good amount of registrations. Additionally > each phone subscribes to itself for MWI and some phone subscribes to > others for BLF. > > Registrar database looks fine. No unused entries there. > > First I will upgrade to recent svn trunk. If that doesn't help, I will > run valgrind on my production system and hope that my machine is strong > enough to deliver its service even with valrgind. > > regards > Helmut > > > On 06.04.2009 16:04, B Karthik wrote: > > It could be due to registrations. I am currently trying to troubleshoot > > this problem. I used a sipp scenario to authenticate with fs and > > register about 2000 different accounts (absolutely no calls made on the > > test setup). Memory usage increases continuously and does not decrease > > at all and crosses more than 1 GB in a few hours. On the other hand, > > there is another fs setup with bypass media turned on and no > > registrations and is up for almost 45 days without restart and has > > consumed only about 95 MB of memory and twice as much virtual memory. > > > > B Karthik > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFJ2hJl4tZeNddg3dwRAmEUAKCTt1aBPlp1pgs3RHw2AVEuH8ixqgCfWkm8 > 6vsgh6Ha34/gdg6iDEEEOR0= > =2H4m > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > View this message in context: > http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593558.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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593704.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Mon Apr 6 08:36:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 6 Apr 2009 10:36:36 -0500 Subject: [Freeswitch-users] How to call multi gateways for failover with early media? In-Reply-To: <1239025636.12559.13.camel@dk-d820> References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com> <20090403235300.GA10045@jdc.jasonjgw.net> <281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org> <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com> <1238987537.13971.13.camel@dk-d820> <2d9149cd0904052108h78dfff26j2a804904e31ba5f3@mail.gmail.com> <1239025636.12559.13.camel@dk-d820> Message-ID: <191c3a030904060836ka14d59ciaa89f1e7a768df8@mail.gmail.com> The default in originate is to return as soon as there is media. So if you bridge an inbound call, FS core will use originate to establish the outbound leg, as soon as it gets media (18X + sdp) it will return and enter the bridge in early media, this allows you to hear the early media while you are waiting for answer. If you want to wait for answer you add {ignore_early_media=true} to the dial string which tells originate to wait for answer or hangup before returning. if you are doing a forked dial and you don't just want the first one that has media to send a 183, you need to also enable {ignore_early_media=true} for that call. On Mon, Apr 6, 2009 at 8:47 AM, David Knell wrote: > On Mon, 2009-04-06 at 00:08 -0400, Kristian Kielhofner wrote: > > > Actually using 180 w/o SDP provides for enhanced call handing > > functionality while only requiring (in many cases) one additional test > > scenario. Consider the current example (all 180s are actually 180s > > w/o SDP and 183 is 183 w/ SDP): > > > > Bridging a call to multiple destinations (A, B, and C). > > > > A: 100,180 > > B: 100,180,200 > > C: 100,183 > > > > We could have implemented proper forking if it weren't for C who > > insisted on sending media early (for whatever reason). While I could > > see many scenarios where this might happen even with the configuration > > I suggest, consider what would happen in the ideal scenario: > > > > A: 100,180 > > B: 100,180,200 > > C: 100,180 > > > Ah, B won because it was the first endpoint to actually /answer/ the > > call and begin playing media. Nice and clean. > > Hang on - if you want to bridge the call on *answer*, then bridge it on > answer, not when one leg starts sending you early media. I've no idea > if FS supports this behaviour for its forked dialling, but it's easy > to do with a bunch of originates, and uuid_bridge the inbound leg to the > first one which answers. > > > People poke at SIP all the time for this one but this is where the > > PSTN even seems a bit ambiguous. We have ISDN cause codes AND inband > > audio messages? > > Yes. A clearing code is used when the call's cleared; inband audio > can be used to give the caller more information than a simple clearing > code might allow - for example, "The number you are calling has been > changed. Please redial on whatever the new number might be." It > makes eminent sense - simple, common causes (e.g. user busy) get dealt > with as part of the call clearing and it's the responsibility of the > originating switch to tell the user; more (indeed arbitrarily) complex > ones are dealt with by the far end. > > --Dave > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/6ff22ae1/attachment-0002.html From codecomplete at free.fr Mon Apr 6 08:56:12 2009 From: codecomplete at free.fr (Fred) Date: Mon, 06 Apr 2009 17:56:12 +0200 Subject: [Freeswitch-users] Freeswitch not started OK at boottime? Message-ID: <7.0.1.0.2.20090406175519.024273d8@fredshack.com> Brian West-3 > What run level are you starting freeswitch? 3 to 5, the default being 5 (it's the desktop version, hence starting with X): # cat /etc/inittab [...] id:5:initdefault: # chkconfig -l freeswitch freeswitch 0:off 1:off 2:off 3:on 4:on 5:on 6:off dujinfang > It seems started, I never used a suse, however, can you try this? #netstat -an | grep 8021 # netstat -an | grep 8021 tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN > Maybe FS started before network is ready. Check scripts in /etc/rc.d/ or any equiv It looks ok: # ll /etc/rc.d/rc5.d/ [...] lrwxrwxrwx 1 root root 10 Jun 16 2008 S05network -> ../network [...] lrwxrwxrwx 1 root root 13 Mar 24 16:36 S12freeswitch -> ../freeswitch lrwxrwxrwx 1 root root 6 Jun 16 2008 S12xdm -> ../xdm lrwxrwxrwx 1 root root 8 Jun 16 2008 S14smbfs -> ../smbfs lrwxrwxrwx 1 root root 13 Jun 16 2008 S15cupsrenice -> ../cupsrenice If it's a Suse-specific issue, I'll go ask in a Suse forum and see if someone can figure it out. Thanks guys for the tips. From carthick84 at gmail.com Mon Apr 6 09:06:01 2009 From: carthick84 at gmail.com (B Karthik) Date: Mon, 6 Apr 2009 09:06:01 -0700 (PDT) Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <1239031841283-2593704.post@n2.nabble.com> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> <1239030142502-2593558.post@n2.nabble.com> <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> <1239031841283-2593704.post@n2.nabble.com> Message-ID: <1239033961503-2593907.post@n2.nabble.com> Great work. Memory usage is constant now. Memory is now Res :162M Virt: 483M for more than 10 mins without increasing. Call rate was set to 100 in sipp. However "top" usage is very high - 137% - 200%. MySQL usage is about 3% constant. I will also try overriding the XML bind function with a function which can lookup an in memory cached hash table for directory entries. I will also try by disabling mysql and post the results shortly. yes, i did exactly as you mentioned. I will try building again from a fresh checkout. I am sorry about not following the policy, I didn't intend to report it as a bug since i was still unsure that it could be a problem in Freeswitch. did you both follow the policy to upgrade? stop fs type make current restart fs if you do not rebuild sofia too (only happens in make current) I just fixed all the problems with these symptoms, 38 million registrations in a 2 day span using 62mb btw, did we not make the policy clear enough about not reporting bugs on the mailing list? On Mon, Apr 6, 2009 at 10:02 AM, B Karthik wrote: > > I updated to the latest revision. No Luck > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > in my scenario I have a reregistration interval of 60 seconds and 32 sip > phones connected. So I have a good amount of registrations. Additionally > each phone subscribes to itself for MWI and some phone subscribes to > others for BLF. > > Registrar database looks fine. No unused entries there. > > First I will upgrade to recent svn trunk. If that doesn't help, I will > run valgrind on my production system and hope that my machine is strong > enough to deliver its service even with valrgind. > > regards > Helmut > > > On 06.04.2009 16:04, B Karthik wrote: > > It could be due to registrations. I am currently trying to troubleshoot > > this problem. I used a sipp scenario to authenticate with fs and > > register about 2000 different accounts (absolutely no calls made on the > > test setup). Memory usage increases continuously and does not decrease > > at all and crosses more than 1 GB in a few hours. On the other hand, > > there is another fs setup with bypass media turned on and no > > registrations and is up for almost 45 days without restart and has > > consumed only about 95 MB of memory and twice as much virtual memory. > > > > B Karthik > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFJ2hJl4tZeNddg3dwRAmEUAKCTt1aBPlp1pgs3RHw2AVEuH8ixqgCfWkm8 > 6vsgh6Ha34/gdg6iDEEEOR0= > =2H4m > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > View this message in context: > http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593558.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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://n2.nabble.com/High-CPU-load-but-only-few-sessions-tp2538703p2593907.html Sent from the freeswitch-users mailing list archive at Nabble.com. From helmut.kuper at ewetel.de Mon Apr 6 09:09:12 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 06 Apr 2009 18:09:12 +0200 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> <1239030142502-2593558.post@n2.nabble.com> <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> Message-ID: <49DA2928.9030205@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Anthony, I did a fresh checkout, compiled it, installed it into a clean directory and will switch over to it tomorrow morning. I hope I can reuse this existing directories: db/ conf/ storage/ sounds/ On 06.04.2009 17:21, Anthony Minessale wrote: > btw, > did we not make the policy clear enough about not reporting bugs on the > mailing list? Yes you did ... I apologize! regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFJ2iko4tZeNddg3dwRAnfWAKCHz7MeJZscWPLMkKQV6lflp8Wi+gCePRQm Cz3J66fWtZEMK+n7D8GXAM8= =mFi1 -----END PGP SIGNATURE----- From andy at fabulous4.co.uk Mon Apr 6 09:10:06 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 6 Apr 2009 17:10:06 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: Message-ID: <517C212F5DE6459181AE16036E04179F@wsandy> Hi Brian, Ok, all up to date, the errors have gone and the software is basically working but the cut off problem still exists. I have an identical software install running on a machine that is not behind a firewall and the cut off doesn't seem to occur. This would seem to suggest it's firewall related. Any clues? regards Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 14:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please update... rebootstrap.. you caught SVN with the libtool patch which kinda broken a few things linking. /b On Apr 6, 2009, at 4:07 AM, Andy Ayers wrote: Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/78947bac/attachment-0002.html From brian at freeswitch.org Mon Apr 6 09:25:46 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Apr 2009 11:25:46 -0500 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <517C212F5DE6459181AE16036E04179F@wsandy> References: <517C212F5DE6459181AE16036E04179F@wsandy> Message-ID: <70CD2E47-AB93-46E7-9198-ED2F5FEBC264@freeswitch.org> Don't record in Mp3, I don't recommend it.. /b On Apr 6, 2009, at 11:10 AM, Andy Ayers wrote: > Hi Brian, > > Ok, all up to date, the errors have gone and the software is > basically working but the cut off problem still exists. I have an > identical software install running on a machine that is not behind a > firewall and the cut off doesn't seem to occur. This would seem to > suggest it's firewall related. Any clues? > > regards > Andy > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/95669636/attachment-0002.html From andy at fabulous4.co.uk Mon Apr 6 09:31:47 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 6 Apr 2009 17:31:47 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: Message-ID: <2D49B31443C64C1AB49E9264C71A5A21@wsandy> Hi Brian, Just doing some more testing, simplified the call by not even trying to record the incoming audio and placing a while (session.ready()) {} loop in the ivr code instead and the calls all now terminate with RECOVERY_ON_TIMER_EXPIRE. Does this shed any light on the subject at all? regards Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 14:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please update... rebootstrap.. you caught SVN with the libtool patch which kinda broken a few things linking. /b On Apr 6, 2009, at 4:07 AM, Andy Ayers wrote: Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/d6ddf6af/attachment-0002.html From brian at freeswitch.org Mon Apr 6 09:39:11 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Apr 2009 11:39:11 -0500 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <2D49B31443C64C1AB49E9264C71A5A21@wsandy> References: <2D49B31443C64C1AB49E9264C71A5A21@wsandy> Message-ID: <581BF129-4E4B-4BBD-9F14-7EB88138101F@freeswitch.org> Three letters come to mind... N A T! ;) What is your network topo? /b On Apr 6, 2009, at 11:31 AM, Andy Ayers wrote: > Hi Brian, > > Just doing some more testing, simplified the call by not even trying > to record the incoming audio and placing a while (session.ready()) > {} loop in the ivr code instead and the calls all now terminate with > RECOVERY_ON_TIMER_EXPIRE. > > Does this shed any light on the subject at all? > > regards > Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/efe51686/attachment-0002.html From mszlazak at aol.com Sun Apr 5 10:14:32 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sun, 05 Apr 2009 13:14:32 -0400 Subject: [Freeswitch-users] MANDATORY_IE_MISSING and NORMAL_TEMPORARY_FAILURE using javascript in dialplan. Message-ID: <8CB84291EE83AD4-D1C-FC0@webmail-db21.sysops.aol.com> On Windows XP/SP3 with FS trunk 12653M I get these errors using javascript in my dialplan: ? [MANDATORY_IE_MISSING] (see pastebin below) and/or with [CS_EXCHANGE_MEDIA] I get [NORMAL_TEMPORARY_FAILURE]? (not shown this time in pastebin) Here is the test javascript file: session.answer(); session.setVariable("choice", "demo"); If I remove session.answer() in the above test script then there is no problem but that doesn't always work with another scripts. Here is the related section of the dialplan. You dial into ext. 2222 and the problem happens at . The variable "choice" is set to "demo" to get here. ? ??? ??? ? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ? ??? ? ??? ??? ??? ??? ??? ??? ??? ? ??? ??? ? ??? ??? ??? ??? ??? ??? ?? ??? ??? ??? ? ??? ??? ? ??? ? ??? ? ??? ??? ? ??? ??? ?? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ?? ??? ??? ??? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ? ??? ? ??? ? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ??? ? ??? ??? ? ??? I enabled SIP/Sofia tracing and "pastebinned" part of the output here: ?http://pastebin.freeswitch.org/8321 Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090405/f000a858/attachment-0002.html From andy at fabulous4.co.uk Mon Apr 6 09:52:51 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 6 Apr 2009 17:52:51 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <581BF129-4E4B-4BBD-9F14-7EB88138101F@freeswitch.org> Message-ID: Hi Brian, The freeswitch server is connect to the internet via a Cisico ASA firewall currently running in NAT mode. I believe it's that simple but can't be sure of the equipment between my firewall and the internet. regards Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 17:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Three letters come to mind... N A T! ;) What is your network topo? /b On Apr 6, 2009, at 11:31 AM, Andy Ayers wrote: Hi Brian, Just doing some more testing, simplified the call by not even trying to record the incoming audio and placing a while (session.ready()) {} loop in the ivr code instead and the calls all now terminate with RECOVERY_ON_TIMER_EXPIRE. Does this shed any light on the subject at all? regards Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/a106bb4b/attachment-0002.html From mike at jerris.com Mon Apr 6 13:03:42 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 6 Apr 2009 16:03:42 -0400 Subject: [Freeswitch-users] libtool 2.2 patch Message-ID: <0CFA8FFA-4621-4133-A0A4-C18EA55BDC99@jerris.com> I need some testers for systems using both libtool 2.2 and 1.5.x to confirm the following patch: http://jira.freeswitch.org/secure/attachment/11356/fs-r12922-libtool22.patch In order to test you will need to do a complete fresh checkout, apply this patch, then do a bootstrap, configure, etc. Please make sure both mod_spidermonkey and mod_shout both build AND load (you will need to edit modules.conf and modules.conf.xml for these modules) when you start freeswitch without any errors. Please post your test results to http://jira.freeswitch.org/browse/FSBUILD-82 Thanks Mike From fax at virgintechnologies.com Mon Apr 6 14:54:05 2009 From: fax at virgintechnologies.com (Justin Miller) Date: Mon, 06 Apr 2009 21:54:05 +0000 Subject: [Freeswitch-users] Native G729 file playback and recording in Windows Message-ID: I know there is an implementation of this for linux. Does anyone have it working in Windows? I gave it a try, but had no luck. I can get individual G729 files to play through the dialplan, but I couldn't get voicemail to work. Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/81a9099d/attachment-0002.html From mszlazak at aol.com Mon Apr 6 15:42:39 2009 From: mszlazak at aol.com (mszlazak) Date: Mon, 6 Apr 2009 15:42:39 -0700 (PDT) Subject: [Freeswitch-users] MANDATORY_IE_MISSING and NORMAL_TEMPORARY_FAILURE using javascript in dialplan. In-Reply-To: <8CB84291EE83AD4-D1C-FC0@webmail-db21.sysops.aol.com> References: <8CB84291EE83AD4-D1C-FC0@webmail-db21.sysops.aol.com> Message-ID: <22918984.post@talk.nabble.com> I svn'd to today's latest trunk to see if the problem remained. However, things seemed to have turned worse. The error messages I had before don't occur but I still can't bridge to my other application with choice=demo. I tried the application by dialing straight into it with the following dial plan. This has worked in FS 1.0.1 through 1.0.3 but fails to make any connection in the current svn trunk. NOTE ON UNRELATED ERROR: I don't use Lua but there was an error in compiling LUA on Windows with 2008 Express so I get a error loading mod_lua.dll today. Simple dialpan that worked before: SIP TRACE: http://pastebin.freeswitch.org/8336 -- View this message in context: http://www.nabble.com/MANDATORY_IE_MISSING-and-NORMAL_TEMPORARY_FAILURE-using-javascript-in-dialplan.-tp22912880p22918984.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Mon Apr 6 15:47:32 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 6 Apr 2009 18:47:32 -0400 Subject: [Freeswitch-users] MANDATORY_IE_MISSING and NORMAL_TEMPORARY_FAILURE using javascript in dialplan. In-Reply-To: <22918984.post@talk.nabble.com> References: <8CB84291EE83AD4-D1C-FC0@webmail-db21.sysops.aol.com> <22918984.post@talk.nabble.com> Message-ID: <8F7C08B5-FF31-4787-877C-E138DCF64E0C@jerris.com> On Apr 6, 2009, at 6:42 PM, mszlazak wrote: > > NOTE ON UNRELATED ERROR: I don't use Lua but there was an error in > compiling > LUA on Windows with 2008 Express so I get a error loading > mod_lua.dll today. This is fixed in svn a bit earlier today. Mike From trevor at concipient.net Mon Apr 6 16:11:31 2009 From: trevor at concipient.net (Trevor Hammonds) Date: Mon, 6 Apr 2009 16:11:31 -0700 Subject: [Freeswitch-users] libtool 2.2 patch In-Reply-To: <0CFA8FFA-4621-4133-A0A4-C18EA55BDC99@jerris.com> References: <0CFA8FFA-4621-4133-A0A4-C18EA55BDC99@jerris.com> Message-ID: <711825c70904061611m5c26c500o2b1cae3c86ed2cb8@mail.gmail.com> When I attempt to apply the patch to rev 12932, it says that the patch is already detected. Has this already been merged? Sincerely, Trevor Hammonds On Mon, Apr 6, 2009 at 1:03 PM, Michael Jerris wrote: > I need some testers for systems using both libtool 2.2 and 1.5.x to > confirm the following patch: > > > http://jira.freeswitch.org/secure/attachment/11356/fs-r12922-libtool22.patch > > In order to test you will need to do a complete fresh checkout, apply > this patch, then do a bootstrap, configure, etc. > > Please make sure both mod_spidermonkey and mod_shout both build AND > load (you will need to edit modules.conf and modules.conf.xml for > these modules) when you start freeswitch without any errors. Please > post your test results to http://jira.freeswitch.org/browse/FSBUILD-82 > > Thanks > Mike > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/3a7b9a7e/attachment-0002.html From diego.viola at gmail.com Mon Apr 6 16:31:05 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 6 Apr 2009 19:31:05 -0400 Subject: [Freeswitch-users] libtool 2.2 patch In-Reply-To: <711825c70904061611m5c26c500o2b1cae3c86ed2cb8@mail.gmail.com> References: <0CFA8FFA-4621-4133-A0A4-C18EA55BDC99@jerris.com> <711825c70904061611m5c26c500o2b1cae3c86ed2cb8@mail.gmail.com> Message-ID: <86a32abc0904061631k530b3ce3j54d489c032f030d2@mail.gmail.com> Hi Trevor, The patch has been merged on latest trunk already. Regards, Diego V. On Mon, Apr 6, 2009 at 7:11 PM, Trevor Hammonds wrote: > When I attempt to apply the patch to rev 12932, it says that the patch is > already detected. Has this already been merged? > > Sincerely, > Trevor Hammonds > > > On Mon, Apr 6, 2009 at 1:03 PM, Michael Jerris wrote: > >> I need some testers for systems using both libtool 2.2 and 1.5.x to >> confirm the following patch: >> >> >> http://jira.freeswitch.org/secure/attachment/11356/fs-r12922-libtool22.patch >> >> In order to test you will need to do a complete fresh checkout, apply >> this patch, then do a bootstrap, configure, etc. >> >> Please make sure both mod_spidermonkey and mod_shout both build AND >> load (you will need to edit modules.conf and modules.conf.xml for >> these modules) when you start freeswitch without any errors. Please >> post your test results to http://jira.freeswitch.org/browse/FSBUILD-82 >> >> Thanks >> 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/20090406/48c0c601/attachment-0002.html From trevor at concipient.net Mon Apr 6 16:42:29 2009 From: trevor at concipient.net (Trevor Hammonds) Date: Mon, 6 Apr 2009 16:42:29 -0700 Subject: [Freeswitch-users] libtool 2.2 patch In-Reply-To: <86a32abc0904061631k530b3ce3j54d489c032f030d2@mail.gmail.com> References: <0CFA8FFA-4621-4133-A0A4-C18EA55BDC99@jerris.com> <711825c70904061611m5c26c500o2b1cae3c86ed2cb8@mail.gmail.com> <86a32abc0904061631k530b3ce3j54d489c032f030d2@mail.gmail.com> Message-ID: <711825c70904061642h27f19572mdf640e95e45b1dda@mail.gmail.com> Thanks! On Mon, Apr 6, 2009 at 4:31 PM, Diego Viola wrote: > Hi Trevor, > > The patch has been merged on latest trunk already. > > Regards, > > Diego V. > > On Mon, Apr 6, 2009 at 7:11 PM, Trevor Hammonds wrote: > >> When I attempt to apply the patch to rev 12932, it says that the patch is >> already detected. Has this already been merged? >> >> Sincerely, >> Trevor Hammonds >> >> >> On Mon, Apr 6, 2009 at 1:03 PM, Michael Jerris wrote: >> >>> I need some testers for systems using both libtool 2.2 and 1.5.x to >>> confirm the following patch: >>> >>> >>> http://jira.freeswitch.org/secure/attachment/11356/fs-r12922-libtool22.patch >>> >>> In order to test you will need to do a complete fresh checkout, apply >>> this patch, then do a bootstrap, configure, etc. >>> >>> Please make sure both mod_spidermonkey and mod_shout both build AND >>> load (you will need to edit modules.conf and modules.conf.xml for >>> these modules) when you start freeswitch without any errors. Please >>> post your test results to http://jira.freeswitch.org/browse/FSBUILD-82 >>> >>> Thanks >>> Mike >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/43b79b99/attachment-0002.html From mattdfong at gmail.com Mon Apr 6 23:52:54 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 7 Apr 2009 13:52:54 +0700 Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? Message-ID: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> I'm doing some outbound dialing, and want to use mod_vmd to detect if a live person picks up or a voicemail picks up. I've read the wiki, and have been playing around with the dialplan implementation and the lua implementation, along with capturing the mod_vmdvmd::beep event. Using the examples on the wiki, I am able to call a number, sleep for 25 seconds, and mod_vmd usually detects a Beep (the answering machine beep right before you are to speak your message). My question is, is there a way to use mod_vmd to detect if an answering machine or human has picked up within the first 1-2 seconds after being answered? If so, can I get an example of how to set this up? my dialplan to test my lua implementation looks like and matt_vmd.lua looks like print ("--matt_vmd.lua START--") local human_detected = false; local voicemail_detected = false; function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '1' and human_detected == false then print('MATT--I detected a HUMAN'); human_detected = true; return "break"; end if type == "event" and voicemail_detected == false then print('MATT--I detected a VOICEMAIL'); voicemail_detected = true; return "break"; end end session:setInputCallback("onInput"); session:execute("vmd"); session:sleep(25000); print ("--matt_vmd.lua FINISHED--") Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/d8d30c16/attachment-0002.html From kristian.kielhofner at gmail.com Tue Apr 7 00:10:40 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 7 Apr 2009 03:10:40 -0400 Subject: [Freeswitch-users] Buzzing when people speak in conference In-Reply-To: <87f2f3b90904030930r2b82a5c3oa9c558b4c5f7052e@mail.gmail.com> References: <87f2f3b90904030930r2b82a5c3oa9c558b4c5f7052e@mail.gmail.com> Message-ID: <2d9149cd0904070010g20eafa1i7bdb513c824c26e6@mail.gmail.com> 2009/4/3 Michael Collins : > On Fri, Apr 3, 2009 at 7:11 AM, Brian West wrote: >> >> Did it sound more like a machine gun? >> /b > > Comfort noise for General Douglas McArthur I guess... > I thought General Norman Scwarzkopf (Stormin' Norman) would have been more appropriate: http://en.wikipedia.org/wiki/Norman_Schwarzkopf,_Jr. Sorry, I just couldn't resist. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From krice at freeswitch.org Tue Apr 7 00:19:21 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 07 Apr 2009 02:19:21 -0500 Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? In-Reply-To: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> Message-ID: Matt, No that?s all mod_vmd does... If you want to do a more advanced analysis of media stream coming from the client mod_amd is available under a commercial license. This does media analysis to determine machine vs humans based on a hand full of metrics that are tunable. Contact me off list for licensing details Ken Rice krice at freeswitch.org From: Matthew Fong Reply-To: Date: Tue, 7 Apr 2009 13:52:54 +0700 To: Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? I'm doing some outbound dialing, and want to use mod_vmd to detect if a live person picks up or a voicemail picks up. I've read the wiki, and have been playing around with the dialplan implementation and the lua implementation, along with capturing the mod_vmd vmd::beep event. Using the examples on the wiki, I am able to call a number, sleep for 25 seconds, and mod_vmd usually detects a Beep (the answering machine beep right before you are to speak your message). My question is, is there a way to use mod_vmd to detect if an answering machine or human has picked up within the first 1-2 seconds after being answered? If so, can I get an example of how to set this up? my dialplan to test my lua implementation looks like and matt_vmd.lua looks like print ("--matt_vmd.lua START--") local human_detected = false; local voicemail_detected = false; function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '1' and human_detected == false then print('MATT--I detected a HUMAN'); human_detected = true; return "break"; end if type == "event" and voicemail_detected == false then print('MATT--I detected a VOICEMAIL'); voicemail_detected = true; return "break"; end end session:setInputCallback("onInput"); session:execute("vmd"); session:sleep(25000); print ("--matt_vmd.lua FINISHED--") 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/f00f0808/attachment-0002.html From mbrancaleoni at voismart.it Tue Apr 7 01:13:09 2009 From: mbrancaleoni at voismart.it (Matteo) Date: Tue, 7 Apr 2009 10:13:09 +0200 (CEST) Subject: [Freeswitch-users] Skype interaction commands on skypiax In-Reply-To: <7b197bef0904020135j6b56662dy5a0dd2862ac4f35d@mail.gmail.com> Message-ID: <573982303.40161239091989637.JavaMail.root@mx.voismart.com> Ciao Giovanni, I suggest to update the startskype.sh script by adding a "su username" statement, in this way: instead of starting skype as echo "myskypeuser xxx" | DISPLAY=:101 /usr/bin/skype --pipelogin & is better to do: su unixusername -c "echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype --pipelogin &" for two reason: you can easily put config into a non-root user AND the startskype.sh will work also if called from init. in fact, a plain echo "myskypeuser xxx" | DISPLAY=:101 /usr/bin/skype --pipelogin & will not work when called from init script, you have to do (even with root) su root -c "echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype --pipelogin &" in any other way skype will not get the user home directory... This is my 2c experience on centos 5.2. regards, matteo. From andy at fabulous4.co.uk Tue Apr 7 06:10:19 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 7 Apr 2009 14:10:19 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <581BF129-4E4B-4BBD-9F14-7EB88138101F@freeswitch.org> Message-ID: <6F855374D4C44C089862BD60E220EB54@wsandy> Hi Brian, Is NAT a known problem? Is there a work around? The messages on the lists seem to imply other folks have this working ok behind NAT firewalls. What's your recommendation for how I should proceed? regards Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 17:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Three letters come to mind... N A T! ;) What is your network topo? /b On Apr 6, 2009, at 11:31 AM, Andy Ayers wrote: Hi Brian, Just doing some more testing, simplified the call by not even trying to record the incoming audio and placing a while (session.ready()) {} loop in the ivr code instead and the calls all now terminate with RECOVERY_ON_TIMER_EXPIRE. Does this shed any light on the subject at all? regards Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/1b22b3f8/attachment-0002.html From jason at jasonjgw.net Tue Apr 7 00:17:06 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 7 Apr 2009 17:17:06 +1000 Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? In-Reply-To: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> References: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> Message-ID: <20090407071706.GA529@jdc.jasonjgw.net> Matthew Fong wrote: > My question is, is there a way to use mod_vmd to detect if an answering > machine or human has picked up within the first 1-2 seconds after being > answered? Probably not. If you have an algorithm in mind that would achieve this with a high degree of reliability, I'm sure the FreeSWITCH developers would be interested in it. However, as far as I know, there is no reliable way to distinguish, for example, my voice as recorded in a voicemail greeting from my voice giving a live greeting after answering a phone call. Think about it. From freeswitch at philstyle.com Mon Apr 6 19:51:39 2009 From: freeswitch at philstyle.com (Drew Ozier) Date: Mon, 6 Apr 2009 22:51:39 -0400 Subject: [Freeswitch-users] Problem listening to PCMU and ul recordings Message-ID: <2388e50e0904061951p1410dc6ayb2b5994009aaaa20@mail.gmail.com> Hi, I've been using the record_session feature and wish to use PCMU or ul audio format, but when I try to play back the audio in either format, it sounds high-pitch and fast as if it is playing back at 2x speed. I looked at the waveform recorded in PCMU and ul versus what it looks like when I record as wav, and it seems like it is only recording every-other sample (which would explain the pitch and speed). My vars.xml is set to PCMU as the global codec pref and the outbound codec pref. I am recording in stereo (one channel per leg of the call), but I'm not messing with any other recording parameters. Incidentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio file, because I'm getting calls off a T1 (actually off of an AudioCodes Mediant 1000 that is converting the T1 to SIP for me), and I'd like to record precisely what is coming off the wire. I'd be happy to send any configuration files, I'm just currently at a loss for how to proceed. Thanks, Drew Ozier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090406/fbf8341b/attachment-0002.html From dschwartz at xconnect.net Tue Apr 7 06:45:53 2009 From: dschwartz at xconnect.net (David Schwartz) Date: Tue, 7 Apr 2009 14:45:53 +0100 Subject: [Freeswitch-users] Can anyone recommend any provisioning tools for use with FS? Message-ID: <062B8EE81F2EC945A577C3EFAE1DD68E99632F@mail.xconnect.net> Preferably GUI based. Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/658f95b4/attachment-0002.html From pawzlion at gmail.com Tue Apr 7 06:46:57 2009 From: pawzlion at gmail.com (David Robinson) Date: Tue, 07 Apr 2009 23:46:57 +1000 Subject: [Freeswitch-users] problems with Faktortel (AU) and multiple DID's and extensions In-Reply-To: References: Message-ID: <49DB5951.2090006@gmail.com> I have been trying to setup 2 DID's to route to 2 extensions but whenever I try it, the second configured DID always routes to the first extension. In my public.xml I have the following: .... rest of file continues here ... While in my default.xml I have this: .. file continues here ... I got my new friend swk to try and diagnose the problem and using ngrep he found with ngrep that the incoming call to the second extension looked like this: U 203.161.130.133:5060 -> 10.0.0.12:5080 INVITE sip:gw+voicepulse at 10.0.0.12:5080;transport=udp SIP/2.0..Via: SIP/2.0/UDP 203.161.130.133:5060;branch=z9hG4bK75f53071;rport..From: "0451282630" ;tag=as555c5b50..To: ..Contact: ..Call-ID: 7 47befb63a2def723e6796294853cc22 at 203.161.130.133..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 07 Apr 2009 08:18:23 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Content-Type: application/sdp..Content-Length: 290....v=0..o=root 1244 12 44 IN IP4 203.161.130.133..s=session..c=IN IP4 203.161.130.133..t=0 0..m=audio 13806 RTP/AVP 18 3 101..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap :3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv.. He says that the INVITE line should have a DNIS (not sure what that is) in that field to indicate which number to route it to but that for some reason, my provider (Faktortel in Australia) is not supplying that information. Does anyone know whether the problem is really at my provider's end or at my end, and if it's at my end, where ? thanks, pawz From dave at 3c.co.uk Tue Apr 7 07:25:17 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 07 Apr 2009 15:25:17 +0100 Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? In-Reply-To: <20090407071706.GA529@jdc.jasonjgw.net> References: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> <20090407071706.GA529@jdc.jasonjgw.net> Message-ID: <1239114317.16460.16.camel@dk-d820> On Tue, 2009-04-07 at 17:17 +1000, Jason White wrote: > Matthew Fong wrote: > > My question is, is there a way to use mod_vmd to detect if an answering > > machine or human has picked up within the first 1-2 seconds after being > > answered? > > Probably not. If you have an algorithm in mind that would achieve this with a > high degree of reliability, I'm sure the FreeSWITCH developers would be > interested in it. However, as far as I know, there is no reliable way to > distinguish, for example, my voice as recorded in a voicemail greeting from my > voice giving a live greeting after answering a phone call. Think about it. The usual way is to measure how long the person who answers the phone speaks for. A person might say "Hello?", "Hello, this is Alice", "Thank you for calling XYZ. How may I direct your call?" Voicemail will usually be longer - "Hi, this is Bob. I'm sorry I can't take your call right now, so please leave me a message after the tone and I'll get back to you as soon as I can." In the first couple of cases above, this would give you an answer - "human" - within the first few seconds of the call. FreeSWITCH will give you TALK (start of speech (or noise)) and NOTALK (end of) events if you enable VAD. --Dave From brian at freeswitch.org Tue Apr 7 07:34:47 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Apr 2009 09:34:47 -0500 Subject: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping? In-Reply-To: <1239114317.16460.16.camel@dk-d820> References: <4256bf830904062352n598423fvd0af3e6abf017f3@mail.gmail.com> <20090407071706.GA529@jdc.jasonjgw.net> <1239114317.16460.16.camel@dk-d820> Message-ID: <15A5B42A-454C-49FF-8895-B8DC2A01019F@freeswitch.org> Or my personal favorite... Congestion tone! /b On Apr 7, 2009, at 9:25 AM, David Knell wrote: > The usual way is to measure how long the person who answers the phone > speaks for. A person might say "Hello?", "Hello, this is Alice", > "Thank you for calling XYZ. How may I direct your call?" Voicemail > will usually be longer - "Hi, this is Bob. I'm sorry I can't take > your call right now, so please leave me a message after the tone and > I'll get back to you as soon as I can." Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/de00912a/attachment-0002.html From freeswitch at philstyle.com Tue Apr 7 07:36:01 2009 From: freeswitch at philstyle.com (Drew Ozier) Date: Tue, 7 Apr 2009 10:36:01 -0400 Subject: [Freeswitch-users] Problem listening to PCMU and ul recordings Message-ID: <2388e50e0904070736i38213405q17e2b18fcf284779@mail.gmail.com> Hi, I've been using the record_session feature and wish to use PCMU or ul audio format, but when I try to play back the audio in either format, it sounds high-pitch and fast as if it is playing back at 2x speed. I looked at the waveform recorded in PCMU and ul versus what it looks like when I record as wav, and it seems like it is only recording every-other sample (which would explain the pitch and speed). My vars.xml is set to PCMU as the global codec pref and the outbound codec pref. I am recording in stereo (one channel per leg of the call), but I'm not messing with any other recording parameters. Incidentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio file, because I'm getting calls off a T1 (actually off of an AudioCodes Mediant 1000 that is converting the T1 to SIP for me), and I'd like to record precisely what is coming off the wire. I'd be happy to send any configuration files, I'm just currently at a loss for how to proceed. Thanks, Drew Ozier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/4cb19225/attachment-0002.html From drew.ozier at gmail.com Tue Apr 7 07:51:20 2009 From: drew.ozier at gmail.com (Drew Ozier) Date: Tue, 7 Apr 2009 10:51:20 -0400 Subject: [Freeswitch-users] Problem listening to PCMU and ul recordings Message-ID: <2388e50e0904070751v606ae262uee0092db66dbcff5@mail.gmail.com> Hi, I've been using the record_session feature and wish to use PCMU or ul audio format, but when I try to play back the audio in either format, it sounds high-pitch and fast as if it is playing back at 2x speed. I looked at the waveform recorded in PCMU and ul versus what it looks like when I record as wav, and it seems like it is only recording every-other sample (which would explain the pitch and speed). My vars.xml is set to PCMU as the global codec pref and the outbound codec pref. I am recording in stereo (one channel per leg of the call), but I'm not messing with any other recording parameters. Incedentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio file, because I'm getting calls off a T1 (actually off of an AudioCodes Mediant 1000 that is converting the T1 to SIP for me), and I'd like to record precicely what is coming off the wire. I'd be happy to send any configuration files, I'm just currently at a loss for how to proceed. Thanks, Drew Ozier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/2ee503a7/attachment-0002.html From gmaruzz at celliax.org Tue Apr 7 09:32:16 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 7 Apr 2009 18:32:16 +0200 Subject: [Freeswitch-users] Skype interaction commands on skypiax In-Reply-To: <573982303.40161239091989637.JavaMail.root@mx.voismart.com> References: <7b197bef0904020135j6b56662dy5a0dd2862ac4f35d@mail.gmail.com> <573982303.40161239091989637.JavaMail.root@mx.voismart.com> Message-ID: <7b197bef0904070932p6464b6f5k226dabbdd0cb6c66@mail.gmail.com> svn commit -m"skypiax: modified configs/startskype.sh to specify which unix user will start the Skype client instance. Thx to mbrancaleoni at voismart.it" Sending configs/startskype.sh Transmitting file data . Committed revision 12937. :-) On Tue, Apr 7, 2009 at 10:13 AM, Matteo wrote: > Ciao Giovanni, > > I suggest to update the startskype.sh script by adding a "su username" statement, > in this way: > > instead of starting skype as > > echo "myskypeuser xxx" | DISPLAY=:101 /usr/bin/skype --pipelogin & > > is better to do: > > su unixusername -c "echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype --pipelogin &" > > for two reason: > you can easily put config into a non-root user > AND > the startskype.sh will work also if called from init. > > in fact, a plain > > echo "myskypeuser xxx" | DISPLAY=:101 /usr/bin/skype --pipelogin & > > will not work when called from init script, > you have to do (even with root) > > su root -c "echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype --pipelogin &" > > in any other way skype will not get the user home directory... > > This is my 2c experience on centos 5.2. > > regards, > matteo. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 7 10:19:16 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 07 Apr 2009 19:19:16 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption Message-ID: <49DB8B14.70400@gmx.net> I want to do the following: Due to missing Softclient with TLS/SRTP support on my Linux laptop (Zoiper is almost there, but not yet with SRTP) I want to install a local FS to listen on a local IP and then communicate via TLS/SRTP to my FS in the Office. As My Laptop has changing IPs (e.g. Ethernet-Cable, WLAN, UMTS) I want FS to listen on 127.0.0.1, connect my local VoIP client (Twinkle) to the local FS and communicate to the FS in my office through the public or LAN IP (maybe via STUN, public IP may change due to change of network connection). 1st Question: Is that possible or is another solution preferrable? 2nd Question: How can I change the amount of memory FS tries to reserve to an absolute minumum (I only have 1 call at a time). Currently it tries to reserve about 360M if I read that right. Best regards Peter From brian at freeswitch.org Tue Apr 7 10:33:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Apr 2009 12:33:06 -0500 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DB8B14.70400@gmx.net> References: <49DB8B14.70400@gmx.net> Message-ID: <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > 1st Question: Is that possible or is another solution preferrable? Just use FreeSWITCH with mod_portaudio. > 2nd Question: How can I change the amount of memory FS tries to > reserve > to an absolute minumum (I only have 1 call at a time). Currently it > tries to reserve about 360M if I read that right. Thats virtual. Look at RES. > > Best regards > Peter Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/91b69506/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Apr 7 15:32:13 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 7 Apr 2009 23:32:13 +0100 Subject: [Freeswitch-users] Hi Load, but calls still perfect Message-ID: Hi Guys, I'm no Linux guru, but today I inadvertently had 1000+ call attempts going through FS, load according to TOP was 16.5. Calls were still absolutely perfect. Can I throw out the rule book on load ? CPU was ~45% on each core. (dual) Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/92b49b11/attachment-0002.html From Prometheus001 at gmx.net Tue Apr 7 15:52:54 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 00:52:54 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> References: <49DB8B14.70400@gmx.net> <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> Message-ID: <49DBD946.5060406@gmx.net> Thanks Brian, what I was actually looking for was to use a standard SIP soft phone with some additional features. I finally manged to make FS listen on 127.0.0.1 the following way: vars.xml internal.xml The rest is standard configuration. Now communication Laptop-internal is UDP on port 5060 and external via TLS on port 5081, so I have no open port 5060 to the internet. Best regards Peter Brian West schrieb: > > On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> 1st Question: Is that possible or is another solution preferrable? > > Just use FreeSWITCH with mod_portaudio. > >> 2nd Question: How can I change the amount of memory FS tries to reserve >> to an absolute minumum (I only have 1 call at a time). Currently it >> tries to reserve about 360M if I read that right. > > Thats virtual. Look at RES. > >> >> Best regards >> Peter > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 7 16:03:13 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 01:03:13 +0200 Subject: [Freeswitch-users] Conference fails with speex codec Message-ID: <49DBDBB1.3090404@gmx.net> I want to use a low bandwidth codec. But whenever I try to use speex I get an error in the conference. We have FS trunk 1288. Switching back to PCMx it works again. Is there any problem with speex and DTMF or with transcoding? Best regards Peter 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2624 sofia_glue_negotiate_sdp() Set Remote Key [1 AES_CM_128_HMAC_SHA1_32 inline:HU0NdX8n18lnRuEKhmJ1O4zSBaolz3wDtOwWIjy8] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:1917 sofia_glue_build_crypto() Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline:r98Rf+0lojftJOPPW9GZon5SZgB6Kg7FsED4cQV3] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[G722:9:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[PCMU:0:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[PCMA:8:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[GSM:3:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[SPEEX:98:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2786 sofia_glue_negotiate_sdp() Substituting codec SPEEX at 20i@16000h 2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error Opening File [/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav] [System error : No such file or directory.] 2009-04-08 00:43:24 [WARNING] mod_conference.c:4799 conference_function() Cannot ask the user for a pin, ending call2009-04-08 00:43:24 [NOTICE] mod_conference.c:4800 conference_function() Hangup sofia/internal/723328 at sip.mydomain.com [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] From msc at freeswitch.org Tue Apr 7 16:35:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Apr 2009 16:35:58 -0700 Subject: [Freeswitch-users] Conference fails with speex codec In-Reply-To: <49DBDBB1.3090404@gmx.net> References: <49DBDBB1.3090404@gmx.net> Message-ID: <87f2f3b90904071635y11d0ccd7gbfab0537cdfbb3d4@mail.gmail.com> > > > 2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error > Opening File > [/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav] > [System error : No such file or directory.] > 2009-04-08 00:43:24 [WARNING] mod_conference.c:4799 > conference_function() Cannot ask the user for a pin, ending This is curious. Do you see this error about the missing file when you use PCMU? -MC > > call2009-04-08 00:43:24 [NOTICE] mod_conference.c:4800 > conference_function() Hangup sofia/internal/723328 at sip.mydomain.com > [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/dad6c3bf/attachment-0002.html From brian at freeswitch.org Tue Apr 7 16:52:12 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Apr 2009 18:52:12 -0500 Subject: [Freeswitch-users] Conference fails with speex codec In-Reply-To: <87f2f3b90904071635y11d0ccd7gbfab0537cdfbb3d4@mail.gmail.com> References: <49DBDBB1.3090404@gmx.net> <87f2f3b90904071635y11d0ccd7gbfab0537cdfbb3d4@mail.gmail.com> Message-ID: <3C6448A5-1D6A-4612-80A2-644CF0BE8F88@freeswitch.org> Chances are he just doesn't have the 16k sound files installed. /b On Apr 7, 2009, at 6:35 PM, Michael Collins wrote: > > 2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error > Opening File > [/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav] > [System error : No such file or directory.] > 2009-04-08 00:43:24 [WARNING] mod_conference.c:4799 > conference_function() Cannot ask the user for a pin, ending > > This is curious. Do you see this error about the missing file when > you use PCMU? > -MC > > > call2009-04-08 00:43:24 [NOTICE] mod_conference.c:4800 > conference_function() Hangup sofia/internal/723328 at sip.mydomain.com > [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/b36df24f/attachment-0002.html From jim at evolutiontel.net Tue Apr 7 17:13:13 2009 From: jim at evolutiontel.net (jim at evolutiontel.net) Date: Wed, 8 Apr 2009 10:13:13 +1000 Subject: [Freeswitch-users] problems with Faktortel (AU) and multiple D ID's and extensions Message-ID: <0cXkljJNrdVq.Y1qLYL1r@smtp.gmail.com> Hi David, Have seen a similar issue reported on whirlpool recently with another provider, essentially if the ITSP does not forward the To: header with the correct terminating DID you will not be able to determine the extention to route the call to. Am I correct in saying you only have one Faktortel account with 2 DIDs attached? Regards, Jim - original message - Subject: [Freeswitch-users] problems with Faktortel (AU) and multiple DID's and extensions From: David Robinson Date: 07/04/2009 13:50 I have been trying to setup 2 DID's to route to 2 extensions but whenever I try it, the second configured DID always routes to the first extension. In my public.xml I have the following: .... rest of file continues here ... While in my default.xml I have this: .. file continues here ... I got my new friend swk to try and diagnose the problem and using ngrep he found with ngrep that the incoming call to the second extension looked like this: U 203.161.130.133:5060 -> 10.0.0.12:5080 INVITE sip:gw+voicepulse at 10.0.0.12:5080;transport=udp SIP/2.0..Via: SIP/2.0/UDP 203.161.130.133:5060;branch=z9hG4bK75f53071;rport..From: "0451282630" ;tag=as555c5b50..To: ..Contact: ..Call-ID: 7 47befb63a2def723e6796294853cc22 at 203.161.130.133..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 07 Apr 2009 08:18:23 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Content-Type: application/sdp..Content-Length: 290....v=0..o=root 1244 12 44 IN IP4 203.161.130.133..s=session..c=IN IP4 203.161.130.133..t=0 0..m=audio 13806 RTP/AVP 18 3 101..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap :3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv.. He says that the INVITE line should have a DNIS (not sure what that is) in that field to indicate which number to route it to but that for some reason, my provider (Faktortel in Australia) is not supplying that information. Does anyone know whether the problem is really at my provider's end or at my end, and if it's at my end, where ? thanks, pawz _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 wiltingtree at gmail.com Tue Apr 7 17:35:56 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Tue, 7 Apr 2009 20:35:56 -0400 Subject: [Freeswitch-users] Two or more simultaneous calls not working Message-ID: Hello, I wrote an application using FreeSWITCH version 1.0.3, with mod_python and a 64 bit box on Red Hat. The app works fine when one person dials in, but when a second person dials in, the first call stops and waits until the second call is finished. It's really strange - if the first call is right in the middle of playing a prompt, it will just stop, and there will be dead air. As soon as the 2nd call hangs up, the prompt for the first call starts playing right where it left off. I previously had FreeSWITCH installed on a 32 bit CentOS box, and this was not happening. Does anybody have any idea what the cause of this could be? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/1c647dbd/attachment-0002.html From john at feith.com Tue Apr 7 17:37:20 2009 From: john at feith.com (John Wehle) Date: Tue, 7 Apr 2009 20:37:20 -0400 (EDT) Subject: [Freeswitch-users] need help getting ISDN talking to Cisco 3845 Message-ID: <200904080037.n380bKih004889@jwlab.FEITH.COM> Our FreeSWITCH setup has an existing T1 using RBS to talk to a digital modem pack in a Cisco 3845. I'm interested in changing from RBS to ISDN. I changed both sides, restart things, and see FreeSWITCH report: 2009-04-07 18:53:15 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:54:40 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:55:36 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:55:45 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:46 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:47 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:48 [NOTICE] Span:0 Q.921() I frame in invalid state ignored I've attached the configs and Cisco debug below. This is using the native ISDN support in FreeSWITCH with a Sangoma A104d on FreeBSD 6.4. I unfortunately don't currently speak ISDN (though I'm starting to pick up a little as a result of this exercise) ... suggestions / hints regarding what's going on and how to resolve it would be welcomed. -- John ------------------------------ wanpipe2.conf ------------------------------- [devices] wanpipe2 = WAN_AFT_TE1, Comment [interfaces] wbg1 = wanpipe2, , TDM_VOICE, Comment [wanpipe2] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 5 PCIBUS = 5 FE_MEDIA = T1 FE_LCODE = B8ZS FE_FRAME = ESF FE_LINE = 2 TE_CLOCK = MASTER TE_REF_CLOCK = 1 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 2 TDMV_DCHAN = 0 TDMV_HW_DTMF = YES [wbg1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES ------------------------------ zaptel.conf --------------------------------- #Sangoma A104 port 2 [slot:5 bus:5 span:2] span=2,0,0,esf,b8zs bchan=25-47 dchan=48 ------------------------------ openzap.conf -------------------------------- [span zt] ; A104D FE 2 1-6 MICA name => Cisco Digital Modem trunk_type => t1 number => 2487 b-channel => 25-47 d-channel => 48 --------------------------- openzap.conf.xml ------------------------------- ------------------------------ Cisco config -------------------------------- controller T1 1/0 framing ESF linecode b8zs cablelength short 220 pri-group timeslots 1-24 interface Serial1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice modem isdn calling-number 2487 no cdp enable ------------------------------ Cisco debug --------------------------------- #show isdn stat Global ISDN Switchtype = primary-ni ISDN Serial1/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x807FFFFF Number of L2 Discards = 2, L2 Session ID = 117 Total Allocated ISDN CCBs = 0 Apr 7 22:53:44.264: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:53:44.264: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:53:44.312: ISDN Se1/0:23 Q921: User RX <- UAf sapi=0 tei=0 Apr 7 22:53:44.312: %CSM-5-PRI: add PRI at 1/0:23 (index 0) Apr 7 22:53:44.312: %ISDN-6-LAYER2UP: Layer 2 for Interface Se1/0:23, TEI 0 cha nged to up Apr 7 22:53:47.268: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:53:47.268: ISDN Se1/0:23 Q921: User TX -> RRf sapi=0 tei=0 nr=0 prepnet-rt# Apr 7 22:53:57.336: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:53:57.336: ISDN Se1/0:23 Q921: User TX -> RRf sapi=0 tei=0 nr=0 prepnet-rt# Apr 7 22:54:11.692: ISDN Se1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0 Apr 7 22:54:11.692: ISDN Se1/0:23 Q921: User TX -> UAf sapi=0 tei=0 Apr 7 22:54:21.760: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:54:21.760: ISDN Se1/0:23 Q921: User TX -> RRf sapi=0 tei=0 nr=0 Apr 7 22:54:51.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:52.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:53.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:54.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:55.760: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:54:55.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:56.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:57.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:58.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:59.760: %CSM-5-PRI: delete PRI at 1/0:23 (index 0) Apr 7 22:54:59.760: %ISDN-6-LAYER2DOWN: Layer 2 for Interface Se1/0:23, TEI 0 c hanged to down Apr 7 22:54:59.760: ISDN Se1/0:23 Q931: Ux_DLRelInd: DL_REL_IND received from L 2 Apr 7 22:55:04.760: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:55:04.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:05.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:05.872: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:55:06.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:07.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:08.760: ISDN Se1/0:23 Q931: Ux_DLRelInd: DL_REL_IND received from L 2 Apr 7 22:55:13.760: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:55:13.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:14.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:15.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:15.772: ISDN Se1/0:23 Q921: User RX <- UAf sapi=0 tei=0 Apr 7 22:55:15.772: %CSM-5-PRI: add PRI at 1/0:23 (index 0) Apr 7 22:55:15.772: %ISDN-6-LAYER2UP: Layer 2 for Interface Se1/0:23, TEI 0 cha nged to up Apr 7 22:55:15.772: ISDN Se1/0:23 Q921: User TX -> INFO sapi=0 tei=0, ns=0 nr=0 Apr 7 22:55:15.772: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:16.772: ISDN Se1/0:23 Q921: S7_T200_EXPIRY: VA = 0, VS = 1 Apr 7 22:55:16.772: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=0 nr= 0 Apr 7 22:55:16.772: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:17.772: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=0 nr= 0 Apr 7 22:55:17.772: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:17.784: ISDN Se1/0:23 Q921: User RX <- RRf sapi=0 tei=0 nr=1 Apr 7 22:55:45.773: ISDN Se1/0:23 Q921: User TX -> INFO sapi=0 tei=0, ns=1 nr=0 Apr 7 22:55:45.773: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:46.773: ISDN Se1/0:23 Q921: S7_T200_EXPIRY: VA = 1, VS = 2 Apr 7 22:55:46.773: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=1 nr= 0 Apr 7 22:55:46.773: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:47.773: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=1 nr= 0 Apr 7 22:55:47.773: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From mszlazak at aol.com Tue Apr 7 18:40:28 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 07 Apr 2009 21:40:28 -0400 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DBD946.5060406@gmx.net> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net> Message-ID: <8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a new trunk and I have to go through vars.xml, etc changing $${local_ip_v4} like you did. Is there a way to change $${local_ip_v4} in one place. That way one wouldn't have remember all the locations that it needs to be changed? -----Original Message----- From: Peter P GMX To: freeswitch-users at lists.freeswitch.org Sent: Tue, 7 Apr 2009 3:52 pm Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption Thanks Brian, what I was actually looking for was to use a standard SIP soft phone with some additional features. I finally manged to make FS listen on 127.0.0.1 the following way: vars.xml internal.xml The rest is standard configuration. Now communication Laptop-internal is UDP on port 5060 and external via TLS on port 5081, so I have no open port 5060 to the internet. Best regards Peter Brian West schrieb: > > On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> 1st Question: Is that possible or is another solution preferrable? > > Just use FreeSWITCH with mod_portaudio. > >> 2nd Question: How can I change the amount of memory FS tries to reserve >> to an absolute minumum (I only have 1 call at a time). Currently it >> tries to reserve about 360M if I read that right. > > Thats virtual. Look at RES. > >> >> Best regards >> Peter > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/7911bc28/attachment-0002.html From jason at jasonjgw.net Tue Apr 7 19:04:56 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 8 Apr 2009 12:04:56 +1000 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> References: <49DBD946.5060406@gmx.net> <8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> Message-ID: <20090408020456.GA11455@jdc.jasonjgw.net> mszlazak at aol.com wrote: > Is there a way to change $${local_ip_v4} in one place. Of course. That's why it's a variable. this goes in vars.xml, substituting the desired address. From tleyden at branchcut.com Tue Apr 7 19:06:27 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Wed, 8 Apr 2009 06:36:27 +0430 Subject: [Freeswitch-users] Two or more simultaneous calls not working Message-ID: > > > Hello, > > I wrote an application using FreeSWITCH version 1.0.3, with mod_python and > a > 64 bit box on Red Hat. > The app works fine when one person dials in, but when a second person dials > in, the first call stops and waits until the second call is finished. It's > really strange - if the first call is right in the middle of playing a > prompt, it will just stop, and there will be dead air. As soon as the 2nd > call hangs up, the prompt for the first call starts playing right where it > left off. > > I previously had FreeSWITCH installed on a 32 bit CentOS box, and this was > not happening. > > Does anybody have any idea what the cause of this could be? > > Thanks! I'm running mod_python fine with 64-bit (Ubuntu 8) and fs svn 12793, and have not seen that problem. Which version of python? Did you build it yourself or was it from a package? If its from a package, please provide the version of Red Hat you are using. One possible cause is that python was not compiled with multi-threading support. I don't know how to check that however .. googled around and didn't find anything. If you fire up your python interpreter and type "import threading" do you get an error? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/bd34aa75/attachment-0002.html From mszlazak at aol.com Tue Apr 7 19:23:41 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 07 Apr 2009 22:23:41 -0400 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <20090408020456.GA11455@jdc.jasonjgw.net> References: <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <20090408020456.GA11455@jdc.jasonjgw.net> Message-ID: <8CB86082B2BA7F3-14E8-2609@FWM-D28.sysops.aol.com> Wonderful! Thank you sir. -----Original Message----- From: Jason White To: freeswitch-users at lists.freeswitch.org Sent: Tue, 7 Apr 2009 7:04 pm Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption mszlazak at aol.com wrote: > Is there a way to change $${local_ip_v4} in one place. Of course. That's why it's a variable. this goes in vars.xml, substituting the desired 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/20090407/564fc12f/attachment-0002.html From solko at gcdf.pl Tue Apr 7 22:54:02 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 07:54:02 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net> <8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> Message-ID: <49DC3BFA.4060200@gcdf.pl> mszlazak at aol.com pisze: > I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > new trunk and I have to go through vars.xml, etc changing > $${local_ip_v4} like you did. > > Is there a way to change $${local_ip_v4} in one place. That way one > wouldn't have remember all the locations that it needs to be changed? > My configuration is not updated when I compile new version and install it. Do you run FS with configuration path pointed to svn work dir? > -----Original Message----- > From: Peter P GMX > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 7 Apr 2009 3:52 pm > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > Thanks Brian, > > > > what I was actually looking for was to use a standard SIP soft phone > > with some additional features. > > > > I finally manged to make FS listen on 127.0.0.1 the following way: > > > > vars.xml > > > > > > internal.xml > > > > > > > > The rest is standard configuration. > > > > Now communication Laptop-internal is UDP on port 5060 and external via > > TLS on port 5081, so I have no open port 5060 to the internet. > > > > Best regards > > Peter > > > > > > > > > > Brian West schrieb: > >> > >> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> 1st Question: Is that possible or is another solution preferrable? > >> > >> Just use FreeSWITCH with mod_portaudio. > >> > >>> 2nd Question: How can I change the amount of memory FS tries to reserve > >>> to an absolute minumum (I only have 1 call at a time). Currently it > >>> tries to reserve about 360M if I read that right. > >> > >> Thats virtual. Look at RES. > >> > >>> > >>> Best regards > >>> Peter > >> > >> Brian West > >> brian at freeswitch.org > > >> > >> -- Meet us a ClueCon! http://www.cluecon.com > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > * > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 mszlazak at aol.com Tue Apr 7 23:43:30 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 08 Apr 2009 02:43:30 -0400 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DC3BFA.4060200@gcdf.pl> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl> Message-ID: <8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> I'm not quite sure what your asking. Are you saying that I could run the latest FS svn but in a way that uses my "older" configuration files? If so then I don't, and don't know how ... blush blush. If that's the easiest thing to do then please tell me how. Thanks. Mark. -----Original Message----- From: Szymon Olko To: freeswitch-users at lists.freeswitch.org Sent: Tue, 7 Apr 2009 10:54 pm Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption mszlazak at aol.com pisze: > I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > new trunk and I have to go through vars.xml, etc changing > $${local_ip_v4} like you did. > > Is there a way to change $${local_ip_v4} in one place. That way one > wouldn't have remember all the locations that it needs to be changed? > My configuration is not updated when I compile new version and install it. Do you run FS with configuration path pointed to svn work dir? > -----Original Message----- > From: Peter P GMX > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 7 Apr 2009 3:52 pm > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > Thanks Brian, > > > > what I was actually looking for was to use a standard SIP soft phone > > with some additional features. > > > > I finally manged to make FS listen on 127.0.0.1 the following way: > > > > vars.xml > > > > > > internal.xml > > > > > > > > The rest is standard configuration. > > > > Now communication Laptop-internal is UDP on port 5060 and external via > > TLS on port 5081, so I have no open port 5060 to the internet. > > > > Best regards > > Peter > > > > > > > > > > Brian West schrieb: > >> > >> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> 1st Question: Is that possible or is another solution preferrable? > >> > >> Just use FreeSWITCH with mod_portaudio. > >> > >>> 2nd Question: How can I change the amount of memory FS tries to reserve > >>> to an absolute minumum (I only have 1 call at a time). Currently it > >>> tries to reserve about 360M if I read that right. > >> > >> Thats virtual. Look at RES. > >> > >>> > >>> Best regards > >>> Peter > >> > >> Brian West > >> brian at freeswitch.org > > >> > >> -- Meet us a ClueCon! http://www.cluecon.com > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > * > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/a1890be2/attachment-0002.html From msc at freeswitch.org Tue Apr 7 23:57:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Apr 2009 23:57:37 -0700 Subject: [Freeswitch-users] need help getting ISDN talking to Cisco 3845 In-Reply-To: <200904080037.n380bKih004889@jwlab.FEITH.COM> References: <200904080037.n380bKih004889@jwlab.FEITH.COM> Message-ID: <87f2f3b90904072357n423feb79mbf206f7fd2dc962d@mail.gmail.com> John, Okay, a few things. First off, the wanpipe2.conf file has a booboo. This line is WRONG: TDMV_DCHAN = 0 For ISDN in North America you want: TDMV_DCHAN = 24 Also, I recommend changing this line: wbg1 = wanpipe2, , TDM_VOICE, Comment To this: wbg1 = wanpipe2, , TDM_VOICE_API, Comment A sample config for Sangoma wanpipeX.conf is here: http://wiki.freeswitch.org/wiki/OpenZAP#Wanpipe_mode Okay, ISDN 101: there is a "network" side and a "user" side. (Also called "terminal" or "cpe"). From what I see here you are trying to have FS be the network side and the Cisco is the user side. Assuming that this is what you want then you will need to use ozmod_libpri because the default OpenZAP PRI stack does not currently support being the network side. You will need to download and install libpri from downloads.digium.com and then you'll need to reconfigure openzap. Follow these instructions to get libpri and openzap working together: http://wiki.freeswitch.org/wiki/OpenZAP#Adding_libpri_Support And then check out this example openzap.conf.xml file for using libpri: http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples#Using_with_PRI_.28libpri_compatibility_stack.29 (Note that you don't want 'cpe' here but rather 'network'.) Now, on the Cisco side... sorry, can't help you. However, I don't see any glaring gotchas from looking at the configs. I don't see where timing is specified nor do I see where the d channel is specified. Hopefully you can confirm that those are set properly. (The cisco needs to be a "slave" to the FS clock, also called "receiving clock"; d-channel is 23 or 24 depending on how cisco numbers their channels.) Have fun! :) -MC On Tue, Apr 7, 2009 at 5:37 PM, John Wehle wrote: > Our FreeSWITCH setup has an existing T1 using RBS to talk to a digital > modem pack in a Cisco 3845. I'm interested in changing from RBS to > ISDN. I changed both sides, restart things, and see FreeSWITCH report: > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090407/695aac05/attachment-0002.html From solko at gcdf.pl Wed Apr 8 00:59:15 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 09:59:15 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl> <8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> Message-ID: <49DC5953.8010000@gcdf.pl> mszlazak at aol.com pisze: > I'm not quite sure what your asking. > Are you saying that I could run the latest FS svn but in a way that uses > my "older" configuration files? If so then I don't, and don't know how > ... blush blush. > If that's the easiest thing to do then please tell me how. > Thanks. Mark. Exactly, I do it that way. For first time I gave installation prefix when configuring FS. You can stay with /usr/local/freeswich/. Now every time i call 'make current' and it does not overwrite my configuration file. In case of huge changes in modules I copy/merge my config file with the one from svn. I did not have problems with it, because developers makes good default values for new configuration options. I don't know which modules do you use, but in ones I use configuration is not changes a lot, there are new options added which does not break old one. make install do not copy configuration files for me if they are already installed, I have that on production server and all test servers. I assume this is correct behavior and I'm not the only one work like that. I looked in Makefile and it tests for config file before installing, so it does not overwrite them. Regards Szymon > > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 7 Apr 2009 10:54 pm > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > mszlazak at aol.com pisze: > >> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > >> new trunk and I have to go through vars.xml, etc changing > >> $${local_ip_v4} like you did. > >> > >> Is there a way to change $${local_ip_v4} in one place. That way one > >> wouldn't have remember all the locations that it needs to be changed? > >> > > My configuration is not updated when I compile new version and install it. Do > > you run FS with configuration path pointed to svn > > work dir? > > > >> -----Original Message----- > >> From: Peter P GMX > > >> To: freeswitch-users at lists.freeswitch.org > >> Sent: Tue, 7 Apr 2009 3:52 pm > >> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> 127.0.0.1 and memory consumption > >> > >> Thanks Brian, > >> > >> > >> > >> what I was actually looking for was to use a standard SIP soft phone > >> > >> with some additional features. > >> > >> > >> > >> I finally manged to make FS listen on 127.0.0.1 the following way: > >> > >> > >> > >> vars.xml > >> > >> > >> > >> > >> > >> internal.xml > >> > >> > >> > >> > >> > >> > >> > >> The rest is standard configuration. > >> > >> > >> > >> Now communication Laptop-internal is UDP on port 5060 and external via > >> > >> TLS on port 5081, so I have no open port 5060 to the internet. > >> > >> > >> > >> Best regards > >> > >> Peter > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> Brian West schrieb: > >> > >>> > >> > >>> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> > >> > >>>> 1st Question: Is that possible or is another solution preferrable? > >> > >>> > >> > >>> Just use FreeSWITCH with mod_portaudio. > >> > >>> > >> > >>>> 2nd Question: How can I change the amount of memory FS tries to reserve > >> > >>>> to an absolute minumum (I only have 1 call at a time). Currently it > >> > >>>> tries to reserve about 360M if I read that right. > >> > >>> > >> > >>> Thats virtual. Look at RES. > >> > >>> > >> > >>>> > >> > >>>> Best regards > >> > >>>> Peter > >> > >>> > >> > >>> Brian West > >> > >>> brian at freeswitch.org > > > ?>> > >> > >>> > >> > >>> -- Meet us a ClueCon! http://www.cluecon.com > >> > >>> > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> > >> > >>> _______________________________________________ > >> > >>> Freeswitch-users mailing list > >> > >>> Freeswitch-users at lists.freeswitch.org > > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> http://www.freeswitch.org > >> > >>> > >> > >> > >> > >> _______________________________________________ > >> > >> Freeswitch-users mailing list > >> > >> Freeswitch-users at lists.freeswitch.org > > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> ------------------------------------------------------------------------ > >> *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > >> * > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Wed Apr 8 02:27:35 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 08 Apr 2009 11:27:35 +0200 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49DA2928.9030205@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <49DA1265.4050907@ewetel.de> <1239030142502-2593558.post@n2.nabble.com> <191c3a030904060821w4926d40bw206cdffd10bf12f6@mail.gmail.com> <49DA2928.9030205@ewetel.de> Message-ID: <49DC6E07.9000307@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Anthony, after one day running the actual trunk things looks much better than before. FS started 24h ago with 129MB VRAM and grows to 136MB VRAM by now. CPU is around 1.3% Thanks for your work! regards helmut On 06.04.2009 18:09, Helmut Kuper wrote: > Hello Anthony, > > I did a fresh checkout, compiled it, installed it into a clean directory > and will switch over to it tomorrow morning. I hope I can reuse this > existing directories: -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFJ3G4H4tZeNddg3dwRAjU7AJ0T9Fl230VfOS00Wbot3A1DTZtBUwCghFHw /CrpYYhdSGmFy+C6RxaIK1A= =yeof -----END PGP SIGNATURE----- From Prometheus001 at gmx.net Wed Apr 8 03:41:22 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 12:41:22 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl> <8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> Message-ID: <49DC7F52.3050301@gmx.net> Simply do a " make current" without a "make samples". That way the conf files in /usr/local/freeswitch/conf remain untouched. I really very seldomly update the conf files. Best regards Peter mszlazak at aol.com schrieb: > I'm not quite sure what your asking. > Are you saying that I could run the latest FS svn but in a way that > uses my "older" configuration files? If so then I don't, and don't > know how ... blush blush. > If that's the easiest thing to do then please tell me how. > Thanks. Mark. > > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 7 Apr 2009 10:54 pm > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > mszlazak at aol.com pisze: > > > I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > > > new trunk and I have to go through vars.xml, etc changing > > > $${local_ip_v4} like you did. > > > > > > Is there a way to change $${local_ip_v4} in one place. That way one > > > wouldn't have remember all the locations that it needs to be changed? > > > > > My configuration is not updated when I compile new version and install it. Do > > you run FS with configuration path pointed to svn > > work dir? > > > > > -----Original Message----- > > > From: Peter P GMX > > > > To: freeswitch-users at lists.freeswitch.org > > > Sent: Tue, 7 Apr 2009 3:52 pm > > > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > > > 127.0.0.1 and memory consumption > > > > > > Thanks Brian, > > > > > > > > > > > > what I was actually looking for was to use a standard SIP soft phone > > > > > > with some additional features. > > > > > > > > > > > > I finally manged to make FS listen on 127.0.0.1 the following way: > > > > > > > > > > > > vars.xml > > > > > > > > > > > > > > > > > > internal.xml > > > > > > > > > > > > > > > > > > > > > > > > The rest is standard configuration. > > > > > > > > > > > > Now communication Laptop-internal is UDP on port 5060 and external via > > > > > > TLS on port 5081, so I have no open port 5060 to the internet. > > > > > > > > > > > > Best regards > > > > > > Peter > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Brian West schrieb: > > > > > >> > > > > > >> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > > > > > >> > > > > > >>> 1st Question: Is that possible or is another solution preferrable? > > > > > >> > > > > > >> Just use FreeSWITCH with mod_portaudio. > > > > > >> > > > > > >>> 2nd Question: How can I change the amount of memory FS tries to reserve > > > > > >>> to an absolute minumum (I only have 1 call at a time). Currently it > > > > > >>> tries to reserve about 360M if I read that right. > > > > > >> > > > > > >> Thats virtual. Look at RES. > > > > > >> > > > > > >>> > > > > > >>> Best regards > > > > > >>> Peter > > > > > >> > > > > > >> Brian West > > > > > >> brian at freeswitch.org > > > ?>> > > > > > >> > > > > > >> -- Meet us a ClueCon! http://www.cluecon.com > > > > > >> > > > > > >> > > > > > >> > > > > > >> ------------------------------------------------------------------------ > > > > > >> > > > > > >> _______________________________________________ > > > > > >> Freeswitch-users mailing list > > > > > >> Freeswitch-users at lists.freeswitch.org > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > >> http://www.freeswitch.org > > > > > >> > > > > > > > > > > > > _______________________________________________ > > > > > > Freeswitch-users mailing list > > > > > > Freeswitch-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > ------------------------------------------------------------------------ > > > *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > > > * > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Wed Apr 8 04:10:34 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 13:10:34 +0200 Subject: [Freeswitch-users] Conference fails with speex codec In-Reply-To: <3C6448A5-1D6A-4612-80A2-644CF0BE8F88@freeswitch.org> References: <49DBDBB1.3090404@gmx.net> <87f2f3b90904071635y11d0ccd7gbfab0537cdfbb3d4@mail.gmail.com> <3C6448A5-1D6A-4612-80A2-644CF0BE8F88@freeswitch.org> Message-ID: <49DC862A.7060402@gmx.net> That was it. I installed the hd sounds and it works now. Thanks. Brian West schrieb: > Chances are he just doesn't have the 16k sound files installed. > > /b > > On Apr 7, 2009, at 6:35 PM, Michael Collins wrote: > >> >> 2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error >> Opening File >> [/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav] >> [System error : No such file or directory.] >> 2009-04-08 00:43:24 [WARNING] mod_conference.c:4799 >> conference_function() Cannot ask the user for a pin, ending >> >> >> This is curious. Do you see this error about the missing file when >> you use PCMU? >> -MC >> >> >> >> call2009-04-08 00:43:24 [NOTICE] mod_conference.c:4800 >> conference_function() Hangup >> sofia/internal/723328 at sip.mydomain.com >> >> [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Wed Apr 8 04:54:50 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 13:54:50 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> References: <49DB8B14.70400@gmx.net> <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> Message-ID: <49DC908A.9010804@gmx.net> I just wanted to know, how much memory overall is consumed by FS inkl. all Libraries (when used on a Netbook with limited memory), so RES does only show a portion of the overall RAM, FS uses incl. libraries. So I did the following: I restarted my laptop and noted the used memory. I deactivated all not needed modules in FS, started FS and noted the used memory. The difference was 24MB. When a call was present (incl. TLS/SRTP), I noted 25M. This is a really low value. Impressive!. Good job done! Best regards Peter Brian West schrieb: > >> 2nd Question: How can I change the amount of memory FS tries to reserve >> to an absolute minumum (I only have 1 call at a time). Currently it >> tries to reserve about 360M if I read that right. > > Thats virtual. Look at RES. > From solko at gcdf.pl Wed Apr 8 05:12:57 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 14:12:57 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DC908A.9010804@gmx.net> References: <49DB8B14.70400@gmx.net> <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DC908A.9010804@gmx.net> Message-ID: <49DC94C9.6000302@gcdf.pl> Peter P GMX pisze: > I just wanted to know, how much memory overall is consumed by FS inkl. > all Libraries (when used on a Netbook with limited memory), so RES does > only show a portion of the overall RAM, FS uses incl. libraries. > > So I did the following: > I restarted my laptop and noted the used memory. > I deactivated all not needed modules in FS, started FS and noted the > used memory. > The difference was 24MB. When a call was present (incl. TLS/SRTP), I > noted 25M. This is a really low value. Impressive!. > Good job done! > Do you use linux based system? Linux don't return memory once used to free, it uses it for disk buffers but it will free it when needed. So probably much part of that system was for disk buffers and are not used by FS any more. I always thought that memory allocated in libraries are included in process which is using them. Where it should be in your opinion? For external services/servers memory is not included in process but this is not the case in FS. Look at RES to know how much memory it uses. > Best regards > Peter > > Brian West schrieb: >>> 2nd Question: How can I change the amount of memory FS tries to reserve >>> to an absolute minumum (I only have 1 call at a time). Currently it >>> tries to reserve about 360M if I read that right. >> Thats virtual. Look at RES. >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 solko at gcdf.pl Wed Apr 8 05:28:11 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 14:28:11 +0200 Subject: [Freeswitch-users] how to set CF_VERBOSE_EVENTS Message-ID: <49DC985B.8080202@gcdf.pl> I track channels via mod_socket_event, I saw in source there is such flag CF_VERBOSE_EVENTS to make all channel related events contain extended data. Is it possible to set it via 'originate' or 'conference xxx dial' commands. This would ease my system. In scenario when I call user which is not registered I get only CHANNEL_DESTROY event but I cannot connect it to my command. I can monitor BACKGROUND_JOB events but in some cases (like answering) it can take time to come. Basically now I need to wait for first event of those two types (CHANNEL_ORIGINATE , BACKGROUND_JOB), if I have extended data in all events I can ignore BACKGROUND_JOB. Regards Szymon From dujinfang at gmail.com Wed Apr 8 07:15:11 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 8 Apr 2009 22:15:11 +0800 Subject: [Freeswitch-users] how to set CF_VERBOSE_EVENTS In-Reply-To: <49DC985B.8080202@gcdf.pl> References: <49DC985B.8080202@gcdf.pl> Message-ID: <5E75EFE2-1653-444C-BA20-6C0BD6335546@gmail.com> There is a dp_tools verbose_events can set that flag, you may try to transfer into a dialplan or use the inline dialplan try this, not tested. > originate sofia/gateway/my_gw/user at domain.com 'verbose_events,playback:foo.wav,echo' inline http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_InlineDialplan On Apr 8, 2009, at 8:28 PM, Szymon Olko wrote: > I track channels via mod_socket_event, I saw in source there is such > flag CF_VERBOSE_EVENTS to make all channel related events > contain extended data. Is it possible to set it via 'originate' or > 'conference xxx dial' commands. > > This would ease my system. In scenario when I call user which is not > registered I get only CHANNEL_DESTROY event but I cannot > connect it to my command. I can monitor BACKGROUND_JOB events but in > some cases (like answering) it can take time to come. > > Basically now I need to wait for first event of those two types > (CHANNEL_ORIGINATE , BACKGROUND_JOB), if I have extended data in > all events I can ignore BACKGROUND_JOB. > > Regards > > Szymon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 carlos.talbot at gmail.com Wed Apr 8 07:38:10 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 8 Apr 2009 09:38:10 -0500 Subject: [Freeswitch-users] FreeSWITCH running on OpenWrt In-Reply-To: <5800526b0904031429s3b1deb4do13ecf3335e18949a@mail.gmail.com> References: <7.0.1.0.2.20090403120452.024273d8@fredshack.com> <5800526b0904031429s3b1deb4do13ecf3335e18949a@mail.gmail.com> Message-ID: <5800526b0904080738t78a10216o7fd78df9f97bad71@mail.gmail.com> Here's the first draft: http://wiki.freeswitch.org/wiki/OpenWrt Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/f9bf968f/attachment-0002.html From tvietduc at yahoo.com Tue Apr 7 19:49:42 2009 From: tvietduc at yahoo.com (to vietduc) Date: Tue, 7 Apr 2009 19:49:42 -0700 (PDT) Subject: [Freeswitch-users] Help about Conference and it's member_id Message-ID: <199683.40252.qm@web38107.mail.mud.yahoo.com> ?Hi! ?I wonder how to get out the member_id of a member in a conference room? Currently, i found that it is increasing 1 each time, so the first one enter the conference room, his/her member_id would be 1, and the second is 2. Sadly, when i close a conference room (all members leave) and re-open it later, the member_id will be increased since the last time instead of begining at 1 as the first time (said, it may be 3, 4 and go on). Why is that (as i think a usual way is that if there is 3 members in a conference room, their member_id should be 1,2,3 or 0,1,2) and is there anyway to get conference's member_id of a member programatically? ?Thanks in advance! ?Duc To From mchlmll at gmail.com Wed Apr 8 03:34:02 2009 From: mchlmll at gmail.com (Michele M) Date: Wed, 8 Apr 2009 12:34:02 +0200 Subject: [Freeswitch-users] How to design my project ? Message-ID: Hi there, I'm quite a newbie about freeswitch. I have an application (IVR) that needs to have endpoints SIP to register,answer the calls and transfer them to the right phones.(I( have my own SIP server).Moreover it needs also a ASR/TTS API' set to communicate with my ASR/TTS engine ( just for example let's assume it is Cepstral). I'd wouldn't want to have freeswitch running and communicate with it to accomplish that but just to use the libfreeswitch library embedded. As I don't know that much about freeswitch can it be done? or just I need to have freeswitch running as a must? Can somebody point me to the right place where to find example of using library embedded (best examples for what I'm trying to do) as I have not found that many? Thanks in advance Miki -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/20fc3e62/attachment-0002.html From wiltingtree at gmail.com Wed Apr 8 08:18:55 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Wed, 8 Apr 2009 11:18:55 -0400 Subject: [Freeswitch-users] Two or more simultaneous calls not Message-ID: Thanks for the response Traun. The version of Python is 2.4.3, and I didn't build it myself, I installed it with yum. The version of Red Hat is 4.1.2-41. "import threading" works fine, so I don't think it's a Python threading issue. The FreeSWITCH version I installed is the freeswitch-1.0.3.tar.gz located at files.freeswitch.org. I didn't make any major changes to the configuration; I enabled Python and set-up the SIP profile, directory and dialplan. No other changes. Any other help would be appreciated, since I really don't know where to look. Thanks, Adam >Date: Wed, 8 Apr 2009 06:36:27 +0430 >From: Traun Leyden >Subject: Re: [Freeswitch-users] Two or more simultaneous calls not > working >To: freeswitch-users at lists.freeswitch.org >Message-ID: > >Content-Type: text/plain; charset="iso-8859-1" > >> >> >> Hello, >> >> I wrote an application using FreeSWITCH version 1.0.3, with mod_python and >> a >> 64 bit box on Red Hat. >> The app works fine when one person dials in, but when a second person dials >> in, the first call stops and waits until the second call is finished. It's >> really strange - if the first call is right in the middle of playing a >> prompt, it will just stop, and there will be dead air. As soon as the 2nd >> call hangs up, the prompt for the first call starts playing right where it >> left off. >> >> I previously had FreeSWITCH installed on a 32 bit CentOS box, and this was >> not happening. >> >> Does anybody have any idea what the cause of this could be? >> >> Thanks! > > >I'm running mod_python fine with 64-bit (Ubuntu 8) and fs svn 12793, and >have not seen that problem. Which version of python? Did you build it >yourself or was it from a package? If its from a package, please provide >the version of Red Hat you are using. > >One possible cause is that python was not compiled with multi-threading >support. I don't know how to check that however .. googled around and >didn't find anything. If you fire up >your python interpreter and type "import threading" do you get an error? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/f5cf95c7/attachment-0002.html From solko at gcdf.pl Wed Apr 8 09:05:42 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 18:05:42 +0200 Subject: [Freeswitch-users] Help about Conference and it's member_id In-Reply-To: <199683.40252.qm@web38107.mail.mud.yahoo.com> References: <199683.40252.qm@web38107.mail.mud.yahoo.com> Message-ID: <49DCCB56.6090502@gcdf.pl> to vietduc pisze: > Hi! > I wonder how to get out the member_id of a member in a conference room? Currently, i found that it is increasing 1 each time, so the first one enter the conference room, his/her member_id would be 1, and the second is 2. Sadly, when i close a conference room (all members leave) and re-open it later, the member_id will be increased since the last time instead of begining at 1 as the first time (said, it may be 3, 4 and go on). Why is that (as i think a usual way is that if there is 3 members in a conference room, their member_id should be 1,2,3 or 0,1,2) and is there anyway to get conference's member_id of a member programatically? > Thanks in advance! > Duc To > First of all members id is increased in FS instance. So you will never get the same id again, unless you reload FS (maybe mod_conference reload also). There is a good reason for that so once you have a member it id stays the same no matter in how many conferences it pass through. All conference commands needs that id so you would be lost if that would change all the time. Yuu did not wrote in which language and how you need it, I'm using mod_event_socket and there listen for CUSTOME_EVENT subevent is conference:maintanace, actions ADDED, DELETED. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 andy at fabulous4.co.uk Wed Apr 8 09:18:23 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Wed, 8 Apr 2009 17:18:23 +0100 Subject: [Freeswitch-users] Using recordFile with Icecast - looses the end of the call Message-ID: <2F161DA684214D6487B8F9711563C62F@wsandy> Hi, I have mod_shout installed and I'm using session.recordFile to capture the audio in a call. When I specify a local file mp3 or wav the audio is captured fine. However, I'm using an icecast server to manage the audio for me and when I specify a remote mp3 location(shout://myserver.com/myaudio.mp3) the end of the call is missing off the resultant mp3 file. A wild shot in the dark I know but does anyone have any experience of this and how it might be resolved? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/a2c3cb19/attachment-0002.html From mszlazak at aol.com Wed Apr 8 09:18:21 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 08 Apr 2009 12:18:21 -0400 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DC5953.8010000@gcdf.pl> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> Message-ID: <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> OK , you're SVN updating on a Linux system but I'm using Windows. The very few times I tried with Tortoise SVN I ran into problems were it would fail because of some path not being present or some strange symbol in a file or something else. Since I'm not experienced enough and don't always have the time, I gave up on this approach and just start over again in a different folder then reconfigure the updated FS and transfer files from an older FS. Yup, it sucks. Thanks.? -----Original Message----- From: Szymon Olko To: freeswitch-users at lists.freeswitch.org Sent: Wed, 8 Apr 2009 12:59 am Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption mszlazak at aol.com pisze: > I'm not quite sure what your asking. > Are you saying that I could run the latest FS svn but in a way that uses > my "older" configuration files? If so then I don't, and don't know how > ... blush blush. > If that's the easiest thing to do then please tell me how. > Thanks. Mark. Exactly, I do it that way. For first time I gave installation prefix when configuring FS. You can stay with /usr/local/freeswich/. Now every time i call 'make current' and it does not overwrite my configuration file. In case of huge changes in modules I copy/merge my config file with the one from svn. I did not have problems with it, because developers makes good default values for new configuration options. I don't know which modules do you use, but in ones I use configuration is not changes a lot, there are new options added which does not break old one. make install do not copy configuration files for me if they are already installed, I have that on production server and all test servers. I assume this is correct behavior and I'm not the only one work like that. I looked in Makefile and it tests for config file before installing, so it does not overwrite them. Regards Szymon > > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 7 Apr 2009 10:54 pm > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > mszlazak at aol.com pisze: > >> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > >> new trunk and I have to go through vars.xml, etc changing > >> $${local_ip_v4} like you did. > >> > >> Is there a way to change $${local_ip_v4} in one place. That way one > >> wouldn't have remember all the locations that it needs to be changed? > >> > > My configuration is not updated when I compile new version and install it. Do > > you run FS with configuration path pointed to svn > > work dir? > > > >> -----Original Message----- > >> From: Peter P GMX > > >> To: freeswitch-users at lists.freeswitch.org > >> Sent: Tue, 7 Apr 2009 3:52 pm > >> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> 127.0.0.1 and memory consumption > >> > >> Thanks Brian, > >> > >> > >> > >> what I was actually looking for was to use a standard SIP soft phone > >> > >> with some additional features. > >> > >> > >> > >> I finally manged to make FS listen on 127.0.0.1 the following way: > >> > >> > >> > >> vars.xml > >> > >> > >> > >> > >> > >> internal.xml > >> > >> > >> > >> > >> > >> > >> > >> The rest is standard configuration. > >> > >> > >> > >> Now communication Laptop-internal is UDP on port 5060 and external via > >> > >> TLS on port 5081, so I have no open port 5060 to the internet. > >> > >> > >> > >> Best regards > >> > >> Peter > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> Brian West schrieb: > >> > >>> > >> > >>> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> > >> > >>>> 1st Question: Is that possible or is another solution preferrable? > >> > >>> > >> > >>> Just use FreeSWITCH with mod_portaudio. > >> > >>> > >> > >>>> 2nd Question: How can I change the amount of memory FS tries to reserve > >> > >>>> to an absolute minumum (I only have 1 call at a time). Currently it > >> > >>>> tries to reserve about 360M if I read that right. > >> > >>> > >> > >>> Thats virtual. Look at RES. > >> > >>> > >> > >>>> > >> > >>>> Best regards > >> > >>>> Peter > >> > >>> > >> > >>> Brian West > >> > >>> brian at freeswitch.org > > > ?>> > >> > >>> > >> > >>> -- Meet us a ClueCon! http://www.cluecon.com > >> > >>> > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> > >> > >>> _______________________________________________ > >> > >>> Freeswitch-users mailing list > >> > >>> Freeswitch-users at lists.freeswitch.org > > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> http://www.freeswitch.org > >> > >>> > >> > >> > >> > >> _______________________________________________ > >> > >> Freeswitch-users mailing list > >> > >> Freeswitch-users at lists.freeswitch.org > > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> ------------------------------------------------------------------------ > >> *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > >> * > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/93853ff1/attachment-0002.html From Prometheus001 at gmx.net Wed Apr 8 09:52:30 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Apr 2009 18:52:30 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DC94C9.6000302@gcdf.pl> References: <49DB8B14.70400@gmx.net> <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DC908A.9010804@gmx.net> <49DC94C9.6000302@gcdf.pl> Message-ID: <49DCD64E.4040104@gmx.net> It's linux, yes. The way I got around the problem that memory may not be freed is: * to reboot the system. * look for used memory * start FS * look for used memory * calculate the difference That way it showed 24-25M which I can understand. Best regards Peter Szymon Olko schrieb: > Peter P GMX pisze: > >> I just wanted to know, how much memory overall is consumed by FS inkl. >> all Libraries (when used on a Netbook with limited memory), so RES does >> only show a portion of the overall RAM, FS uses incl. libraries. >> >> So I did the following: >> I restarted my laptop and noted the used memory. >> I deactivated all not needed modules in FS, started FS and noted the >> used memory. >> The difference was 24MB. When a call was present (incl. TLS/SRTP), I >> noted 25M. This is a really low value. Impressive!. >> Good job done! >> >> > Do you use linux based system? Linux don't return memory once used to free, it uses it for disk buffers but it will free it when > needed. So probably much part of that system was for disk buffers and are not used by FS any more. > > I always thought that memory allocated in libraries are included in process which is using them. Where it should be in your > opinion? For external services/servers memory is not included in process but this is not the case in FS. Look at RES to know how > much memory it uses. > > >> Best regards >> Peter >> >> Brian West schrieb: >> >>>> 2nd Question: How can I change the amount of memory FS tries to reserve >>>> to an absolute minumum (I only have 1 call at a time). Currently it >>>> tries to reserve about 360M if I read that right. >>>> >>> Thats virtual. Look at RES. >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 gmaruzz at celliax.org Wed Apr 8 09:56:37 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 8 Apr 2009 18:56:37 +0200 Subject: [Freeswitch-users] How to design my project ? In-Reply-To: References: Message-ID: <7b197bef0904080956h190a0bb1v6fda1c6965931390@mail.gmail.com> Ciao Michele, as a start is definitely better (and more gratifying) that you runs FreeSWITCH. Then, if (and only if) there is a compelling reason that justify the amount of time needed to develop a standalone application, go for it. Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Apr 8, 2009 at 12:34 PM, Michele M wrote: > Hi there, > > I'm quite a newbie about freeswitch. I have an? application? (IVR) that > needs to have endpoints SIP to register,answer the calls and transfer them > to the right phones.(I( have my own SIP server).Moreover it needs also a > ASR/TTS API' set? to communicate with my ASR/TTS engine ( just for example > let's assume it is Cepstral). I'd wouldn't want to have freeswitch running > and communicate with it to accomplish that but just to use the libfreeswitch > library embedded. As I don't know that much about freeswitch can it be done? > or just I need to have freeswitch running as a must? Can somebody point me > to the right place where to find example of using library embedded (best > examples for what I'm trying to do) as I have not found that many? > > Thanks in advance > > Miki > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 solko at gcdf.pl Wed Apr 8 10:11:29 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 19:11:29 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DCD64E.4040104@gmx.net> References: <49DB8B14.70400@gmx.net> <3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DC908A.9010804@gmx.net> <49DC94C9.6000302@gcdf.pl> <49DCD64E.4040104@gmx.net> Message-ID: <49DCDAC1.70408@gcdf.pl> Peter P GMX pisze: > It's linux, yes. > The way I got around the problem that memory may not be freed is: > > * to reboot the system. > * look for used memory > * start FS > * look for used memory > * calculate the difference > > That way it showed 24-25M which I can understand. > I meant that FS can use less memory now they 24-25 M, what is RES shows is exactly that value. In those 24-25 M are buffers for files which now are not needed for FS and kernel handles that memory. It is show as used but is not used by FS. Kernel uses it and will free it when there will be lack of memory. Always look at RES value if you want to know FS consumption. Szymon > Best regards > Peter > > Szymon Olko schrieb: >> Peter P GMX pisze: >> >>> I just wanted to know, how much memory overall is consumed by FS inkl. >>> all Libraries (when used on a Netbook with limited memory), so RES does >>> only show a portion of the overall RAM, FS uses incl. libraries. >>> >>> So I did the following: >>> I restarted my laptop and noted the used memory. >>> I deactivated all not needed modules in FS, started FS and noted the >>> used memory. >>> The difference was 24MB. When a call was present (incl. TLS/SRTP), I >>> noted 25M. This is a really low value. Impressive!. >>> Good job done! >>> >>> >> Do you use linux based system? Linux don't return memory once used to free, it uses it for disk buffers but it will free it when >> needed. So probably much part of that system was for disk buffers and are not used by FS any more. >> >> I always thought that memory allocated in libraries are included in process which is using them. Where it should be in your >> opinion? For external services/servers memory is not included in process but this is not the case in FS. Look at RES to know how >> much memory it uses. >> >> >>> Best regards >>> Peter >>> >>> Brian West schrieb: >>> >>>>> 2nd Question: How can I change the amount of memory FS tries to reserve >>>>> to an absolute minumum (I only have 1 call at a time). Currently it >>>>> tries to reserve about 360M if I read that right. >>>>> >>>> Thats virtual. Look at RES. >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 solko at gcdf.pl Wed Apr 8 10:36:37 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 19:36:37 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> Message-ID: <49DCE0A5.6080905@gcdf.pl> mszlazak at aol.com pisze: > OK , you're SVN updating on a Linux system but I'm using Windows. The > very few times I tried with Tortoise SVN I ran into problems were it > would fail because of some path not being present or some strange symbol > in a file or something else. Since I'm not experienced enough and don't > always have the time, I gave up on this approach and just start over > again in a different folder then reconfigure the updated FS and transfer > files from an older FS. Yup, it sucks. > Yes I'm linux user. If you have problems with svn update then you can do your way, make fresh checkout every time. After CO copy modules.conf and build new version, just copy old config files to installation directory if it is always different one. That's why I hate gui tools for things like full svn update. > > > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, 8 Apr 2009 12:59 am > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > mszlazak at aol.com pisze: > >> I'm not quite sure what your asking. > >> Are you saying that I could run the latest FS svn but in a way that uses > >> my "older" configuration files? If so then I don't, and don't know how > >> ... blush blush. > >> If that's the easiest thing to do then please tell me how. > >> Thanks. Mark. > > > > Exactly, I do it that way. > > For first time I gave installation prefix when configuring FS. You can stay with > > /usr/local/freeswich/. > > Now every time i call 'make current' and it does not overwrite my configuration > > file. > > > > In case of huge changes in modules I copy/merge my config file with the one from > > svn. I did not have problems with it, because > > developers makes good default values for new configuration options. > > > > I don't know which modules do you use, but in ones I use configuration is not > > changes a lot, there are new options added which > > does not break old one. > > > > make install do not copy configuration files for me if they are already > > installed, I have that on production server and all test > > servers. I assume this is correct behavior and I'm not the only one work like > > that. > > I looked in Makefile and it tests for config file before installing, so it does > > not overwrite them. > > > > Regards > > Szymon > >> > >> -----Original Message----- > >> From: Szymon Olko > > >> To: freeswitch-users at lists.freeswitch.org > >> Sent: Tue, 7 Apr 2009 10:54 pm > >> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> 127.0.0.1 and memory consumption > >> > >> mszlazak at aol.com > pisze: > >> > >>> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > >> > >>> new trunk and I have to go through vars.xml, etc changing > >> > >>> $${local_ip_v4} like you did. > >> > >>> > >> > >>> Is there a way to change $${local_ip_v4} in one place. That way one > >> > >>> wouldn't have remember all the locations that it needs to be changed? > >> > >>> > >> > >> My configuration is not updated when I compile new version and install it. Do > >> > >> you run FS with configuration path pointed to svn > >> > >> work dir? > >> > >> > >> > >>> -----Original Message----- > >> > >>> From: Peter P GMX >> > >> > >>> To: freeswitch-users at lists.freeswitch.org > > >> > >>> Sent: Tue, 7 Apr 2009 3:52 pm > >> > >>> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> > >>> 127.0.0.1 and memory consumption > >> > >>> > >> > >>> Thanks Brian, > >> > >>> > >> > >>> > >> > >>> > >> > >>> what I was actually looking for was to use a standard SIP soft phone > >> > >>> > >> > >>> with some additional features. > >> > >>> > >> > >>> > >> > >>> > >> > >>> I finally manged to make FS listen on 127.0.0.1 the following way: > >> > >>> > >> > >>> > >> > >>> > >> > >>> vars.xml > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> internal.xml > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> The rest is standard configuration. > >> > >>> > >> > >>> > >> > >>> > >> > >>> Now communication Laptop-internal is UDP on port 5060 and external via > >> > >>> > >> > >>> TLS on port 5081, so I have no open port 5060 to the internet. > >> > >>> > >> > >>> > >> > >>> > >> > >>> Best regards > >> > >>> > >> > >>> Peter > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> Brian West schrieb: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> 1st Question: Is that possible or is another solution preferrable? > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Just use FreeSWITCH with mod_portaudio. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> 2nd Question: How can I change the amount of memory FS tries to reserve > >> > >>> > >> > >>>>> to an absolute minumum (I only have 1 call at a time). Currently it > >> > >>> > >> > >>>>> tries to reserve about 360M if I read that right. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Thats virtual. Look at RES. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>>> Best regards > >> > >>> > >> > >>>>> Peter > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Brian West > >> > >>> > >> > >>>> brian at freeswitch.org > > > ?>> > > ?> > >> > >> ?>?>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> -- Meet us a ClueCon! http://www.cluecon.com > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> ------------------------------------------------------------------------ > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> _______________________________________________ > >> > >>> > >> > >>>> Freeswitch-users mailing list > >> > >>> > >> > >>>> Freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>>> http://www.freeswitch.org > >> > >>> > >> > >>>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> _______________________________________________ > >> > >>> > >> > >>> Freeswitch-users mailing list > >> > >>> > >> > >>> Freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>> http://www.freeswitch.org > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > >> > >>> * > > > >> > >>> > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> > >> > >>> _______________________________________________ > >> > >>> Freeswitch-users mailing list > >> > >>> Freeswitch-users at lists.freeswitch.org > > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> _______________________________________________ > >> > >> Freeswitch-users mailing list > >> > >> Freeswitch-users at lists.freeswitch.org > > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> ------------------------------------------------------------------------ > >> New Deals on Dell Netbooks - Now starting at $299 > >> > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 8 10:44:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Apr 2009 12:44:04 -0500 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DCE0A5.6080905@gcdf.pl> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> Message-ID: <3BBB82A7-FA78-4FDB-A44B-672B942C66E5@freeswitch.org> You know if you keep doing a fresh checkout every single time then you are wasting bandwidth... if its your only choice then do that but I highly recommend you learn to use the tools properly. Our bandwidth is kindly provided by Bandwidth.com and I would hate to just waste it for no reason.... btw don't forget to register for Cluecon its quickly approaching. /b On Apr 8, 2009, at 12:36 PM, Szymon Olko wrote: > Yes I'm linux user. > If you have problems with svn update then you can do your way, make > fresh checkout every time. After CO copy modules.conf and > build new version, just copy old config files to installation > directory if it is always different one. > > That's why I hate gui tools for things like full svn update. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/db4ed547/attachment-0002.html From mszlazak at aol.com Wed Apr 8 10:58:31 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 08 Apr 2009 13:58:31 -0400 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DCE0A5.6080905@gcdf.pl> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl><8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> Message-ID: <8CB868AC2870B10-8D4-75@MBLK-M41.sysops.aol.com> Na jaki? czas, b?d? uczy? si?, jak radzi? sobie z Tortoise SVN b??d?w. M?j polski nie jest zbyt dobre, ale dzi?kuj?. Google pomaga. -----Original Message----- From: Szymon Olko To: freeswitch-users at lists.freeswitch.org Sent: Wed, 8 Apr 2009 10:36 am Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption mszlazak at aol.com pisze: > OK , you're SVN updating on a Linux system but I'm using Windows. The > very few times I tried with Tortoise SVN I ran into problems were it > would fail because of some path not being present or some strange symbol > in a file or something else. Since I'm not experienced enough and don't > always have the time, I gave up on this approach and just start over > again in a different folder then reconfigure the updated FS and transfer > files from an older FS. Yup, it sucks. > Yes I'm linux user. If you have problems with svn update then you can do your way, make fresh checkout every time. After CO copy modules.conf and build new version, just copy old config files to installation directory if it is always different one. That's why I hate gui tools for things like full svn update. > > > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, 8 Apr 2009 12:59 am > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > =0 A> mszlazak at aol.com pisze: > >> I'm not quite sure what your asking. > >> Are you saying that I could run the latest FS svn but in a way that uses > >> my "older" configuration files? If so then I don't, and don't know how > >> ... blush blush. > >> If that's the easiest thing to do then please tell me how. > >> Thanks. Mark. > > > > Exactly, I do it that way. > > For first time I gave installation prefix when configuring FS. You can stay with > > /usr/local/freeswich/. > > Now every time i call 'make current' and it does not overwrite my configuration > > file. > > > > In case of huge changes in modules I copy/merge my config file with the one from > > svn. I did not have problems with it, because > > developers makes good default values for new configuration options. > > > > I don't know which modules do you use, but in ones I use configuration is not > > changes a lot, there are new options added which > > does not break old one. > > > > make install do not copy configuration files for me if they are already > > installed, I have that on production server and all test > > servers. I assume this is correct behavior and I'm not the only one work like > > that. > > I looked in Makefile and it tests for config file before installing, so it does > > not overwrite them. > > > > Regards > > Szymon > >>=2 0 > >> -----Original Message----- > >> From: Szymon Olko > > >> To: freeswitch-users at lists.freeswitch.org > >> Sent: Tue, 7 Apr 2009 10:54 pm > >> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> 127.0.0.1 and memory consumption > >> > >> mszlazak at aol.com > pisze: > >> > >>> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > >> > >>> new trunk and I have to go through vars.xml, etc changing > >> > >>> $${local_ip_v4} like you did. > >> > >>> > >> > >>> Is there a way to change $${local_ip_v4} in one place. That way one > >> > >>> wouldn't have remember all the locations that it needs to be changed? > >> > >>> > >> > >> My configuration is not updated when I compile new version and install it. Do > >> > >> you run FS with configuration path pointed to svn > >> > >> work dir? > >> > >> > >> > >>> -----Original Message----- > >> > >>> From: Peter P GMX >> > >> > >>> To: freeswitch-users at lists.freeswitch.org > > >> > >>> Sent: Tue, 7 Apr 2009 3:52 pm > >> > >>> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> > >>> 127.0.0.1 and memory consumption > >> > >>> > >> > >>> Thanks Brian, > >> > >>> > >> > >>> > >> > >>> > >> > >>> what I was actually looking for was to use a standard SIP soft phone > >> > >>> > >> > >>> with some additional features. > >> > >>> > >> > >>> > >> > >>> > >> > >>> I finally manged to make FS listen on 127.0.0.1 the following way: > >> > >>> > >> > >>> > >> > >>> > >> > >>> vars.xml > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> internal.xml > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> The rest is standard configuration. > >> > >>> > >> > >>> > >> > >>> > >> > >>> Now communication Laptop-internal is UDP on port 5060 and external via > >> > >>> > >> > >>> TLS on port 5081, so I have no open port 5060 to the internet. > >> > >>> > >> > >>> > >> > >>> > >> > >>> Best regards > >> > >>> > >> > >>> Peter > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> Brian West schrieb: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> 1st Question: Is that possible or is another solution preferrable? > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Just use FreeSWITCH with mod_portaudio. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> 2nd Question: How can I change the amount of memory FS tries to reserve > >> > >>> > >> > >>>>> to an absolute minumum (I only have 1 call at a time). Currently it > >> > >>> > >> > >>>>> tries to reserve about 360M if I read that right. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Thats virtual. Look at RES. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>>> Best regards > >> > >>> > >> > >>>>> Peter > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Brian West > >> > >>> > >> > >>>> brian at freeswitch.org > > > ?>> > > ?> > >> > >> ?>?>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> -- Meet us a ClueCon! http://www.cluecon.com > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> ------------------------------------------------------------------------ > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> _______________________________________________ > >> > >>> > >> > >>>> Freeswitch-users mailing list > >> > >>> > >> > >>>> Freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswi tch-users > >> > >>> > >> > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>>> http://www.freeswitch.org > >> > >>> > >> > >>>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> _______________________________________________ > >> > >>> > >> > >>> Freeswitch-users mailing list > >> > >>> > >> > >>> Freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>> http://www.freeswitch.org > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > >> > >>> * > > > >> > >>> > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> > >> > >>> _______________________________________________ > >> > >>> Freeswitch-users mailing list > >> > >>> Freeswitch-users at lists.freeswitch.org > > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> _______________________________________________ > >> > >> Freeswitch-users mailing list > >> > >> Freeswitch-users at lists.freeswitch.org > > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> ------------------------------------------------------------------------ > >> New Deals on Dell Netbooks - Now starting at $299 > >> > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/fr eeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/5b019a7e/attachment-0002.html From gkuri at ieee.org Wed Apr 8 11:08:53 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Wed, 08 Apr 2009 11:08:53 -0700 Subject: [Freeswitch-users] rtp/one way audio problem Message-ID: <49DCE835.8030206@ieee.org> We're seeing occasional one way audio issues for international calls going out to one of several carriers. On roughly 2 out of 5 calls outbound, there is no audio on the the calling party's side, however the called party indicates they can hear the calling party perfectly well. NAT is not involved anywhere on our side. Sniffing the traffic shows the rtp stream tries to start, coming in from the carrier, but then stops, which is probably why there's no audio on the calling party's side. There is however, rtp going out from us to the carrier, which is probably why the called party hears the calling party OK. I enabled proxy_media=true and that seems to have fixed the problem (or the problem coincidentally stopped), so now I'm beginning to wonder if we're hitting any of the goofy Sonus bugs described here ... http://wiki.freeswitch.org/wiki/RTP_Issues I know my carrier uses a Cisco, but I also know their routes are transit, not direct routes, so I have no idea what other softswitches the rtp is going through before it finally hits the PSTN in whatever country is being called. Is there anyway to know if there's a Sonus in the media stream somewhere? Do the Cisco's do anything goofy that I need to be aware of as well? I can turn off proxy_media and run a pcapsipdump if that will help? As a side note, we have absolutely no problem terminating domestic calls via any of our domestic carriers/CLECs. This problem seems to be only plagued with International calling. Thanks, Gabe From solko at gcdf.pl Wed Apr 8 11:14:07 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 20:14:07 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <3BBB82A7-FA78-4FDB-A44B-672B942C66E5@freeswitch.org> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> <3BBB82A7-FA78-4FDB-A44B-672B942C66E5@freeswitch.org> Message-ID: <49DCE96F.5070500@gcdf.pl> Brian West pisze: > You know if you keep doing a fresh checkout every single time then you > are wasting bandwidth... if its your only choice then do that but I > highly recommend you learn to use the tools properly. Our bandwidth is > kindly provided by Bandwidth.com and I would hate to just waste it for > no reason.... btw don't forget to register for Cluecon its quickly > approaching. > > /b > Your right about bandwidth, I use svn in console and never had problems that cannot be fixed. Those gui tools they try to be to intelligent. I thought there was console svn tool for windows. Regarding Cluecon, I would like to meet you all there, but in this year it's to expensive and too far for me. Szymon > On Apr 8, 2009, at 12:36 PM, Szymon Olko wrote: > >> Yes I'm linux user. >> If you have problems with svn update then you can do your way, make >> fresh checkout every time. After CO copy modules.conf and >> build new version, just copy old config files to installation >> directory if it is always different one. >> >> That's why I hate gui tools for things like full svn update. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 8 11:23:16 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Apr 2009 13:23:16 -0500 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <49DCE96F.5070500@gcdf.pl> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> <3BBB82A7-FA78-4FDB-A44B-672B942C66E5@freeswitch.org> <49DCE96F.5070500@gcdf.pl> Message-ID: Where are you? /b On Apr 8, 2009, at 1:14 PM, Szymon Olko wrote: > Regarding Cluecon, I would like to meet you all there, but in this > year it's to expensive and too far for me. > > Szymon Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/c9ebb066/attachment-0002.html From brian at freeswitch.org Wed Apr 8 11:23:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Apr 2009 13:23:50 -0500 Subject: [Freeswitch-users] rtp/one way audio problem In-Reply-To: <49DCE835.8030206@ieee.org> References: <49DCE835.8030206@ieee.org> Message-ID: <1E09FF26-77A5-4D4B-8ACF-76909B393E19@freeswitch.org> Do you have any reason to be doing proxy media? /b On Apr 8, 2009, at 1:08 PM, Gabriel Kuri wrote: > I can turn off proxy_media and run a pcapsipdump if that will help? Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/2f6cb535/attachment-0002.html From gkuri at ieee.org Wed Apr 8 11:30:26 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Wed, 08 Apr 2009 11:30:26 -0700 Subject: [Freeswitch-users] rtp/one way audio problem In-Reply-To: <1E09FF26-77A5-4D4B-8ACF-76909B393E19@freeswitch.org> References: <49DCE835.8030206@ieee.org> <1E09FF26-77A5-4D4B-8ACF-76909B393E19@freeswitch.org> Message-ID: <49DCED42.6070606@ieee.org> Brian West wrote: > Do you have any reason to be doing proxy media? no, not other than to fix the one way audio issue :) I'd rather leave proxy_media off. Gabe From solko at gcdf.pl Wed Apr 8 11:35:13 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 08 Apr 2009 20:35:13 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl> <8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> <3BBB82A7-FA78-4FDB-A44B-672B942C66E5@freeswitch.org> <49DCE96F.5070500@gcdf.pl> Message-ID: <49DCEE61.8090802@gcdf.pl> Brian West pisze: > Where are you? > Poland, Wroc?aw. http://maps.google.pl/maps?f=q&source=s_q&hl=pl&geocode=&q=poland,+wroc%C5%82aw&sll=52.025459,19.204102&sspn=6.979078,18.017578&ie=UTF8&ll=51.107833,17.038422&spn=0.222023,0.563049&z=11 > /b > > On Apr 8, 2009, at 1:14 PM, Szymon Olko wrote: > >> Regarding Cluecon, I would like to meet you all there, but in this >> year it's to expensive and too far for me. >> >> Szymon > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 can_man at gmx.de Wed Apr 8 14:24:39 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Wed, 08 Apr 2009 23:24:39 +0200 Subject: [Freeswitch-users] speex can't find OggS header Message-ID: <20090408212439.268950@gmx.net> Hello everyone, I am trying to get a Java Sip client working with speex/16000. FS sets the codec correctly and then starts sending packets to my client: 2009-04-08 21:46:34 [DEBUG] sofia_glue.c:2732 sofia_glue_negotiate_sdp() Audio Codec Compare [speex:100:16000:0]/[SPEEX:99:16000:20] 2009-04-08 21:46:34 [DEBUG] sofia_glue.c:1857 sofia_glue_tech_set_codec() Set Codec sofia/external5090/puli at 97.101.59.118:5090 SPEEX/16000 20 ms 320 samples When the packets arrive jspeex can't decode them and I started to look at them manually to find out what the problem is. The payload of each RTP packet is 42 bytes and when looking for the "OggS" header I can't find it. Or is the ogg header not needed? Jspeex looks for it and as it can't find it, it stops decoding. Is FS sending one frame per packet? Thank you very much for your help. Best wishes, Phil Ps: Wireshark tells me the following for a sample package: Real-Time Transport Protocol Setup Method: SDP 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: speex (100) Sequence number: 9365 Extended sequence number: 74901 Timestamp: 23680 Synchronization Source identifier: 0x004a235c (4858716) Payload: 2C6679456EE347FFD0A3D55B133771A9A100DB639F5B8ED2... Payload is 42 bytes. -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01 From tleyden at branchcut.com Wed Apr 8 15:06:08 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Thu, 9 Apr 2009 02:36:08 +0430 Subject: [Freeswitch-users] Two or more simultaneous calls not Message-ID: Hi Adam, I'm stumped .. I guess you could try the following: * Try with the trunk version of freeswitch. I don't think it will matter, but just in case * Try to simulate the same test with a Lua script. Do you see the same problem? If those don't turn up anything, then the next logical step would be to start adding printf() statements in the mod_python code and find out where it is getting stuck. In particular around the parts where it swaps the threadstate in and out. I might be able to create a patch for you, but try those other tests first. HTH, Traun > > Thanks for the response Traun. The version of Python is 2.4.3, and I > didn't > build it myself, I installed it with yum. > The version of Red Hat is 4.1.2-41. > "import threading" works fine, so I don't think it's a Python threading > issue. > The FreeSWITCH version I installed is the > freeswitch-1.0.3.tar.gz< > http://files.freeswitch.org/freeswitch-1.0.3.tar.gz> > located > at files.freeswitch.org. > I didn't make any major changes to the configuration; I enabled Python and > set-up the SIP profile, directory and dialplan. No other changes. > Any other help would be appreciated, since I really don't know where to > look. > > Thanks, > Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/15f8d417/attachment-0002.html From john at feith.com Wed Apr 8 15:19:00 2009 From: john at feith.com (John Wehle) Date: Wed, 8 Apr 2009 18:19:00 -0400 (EDT) Subject: [Freeswitch-users] need help getting ISDN talking to Cisco 3845 Message-ID: <200904082219.n38MJ0ld006139@jwlab.FEITH.COM> > Okay, a few things. First off, the wanpipe2.conf file has a booboo. Don't think so. > This line is WRONG: > TDMV_DCHAN = 0 Not exactly. My understanding is you can use either: wanpipeX.conf: TDMV_DCHAN = 0 zaptel.conf: dchan = 24 (or in our case 48 since it's the second span) which means use zaptel to handle the d-channel hdlc or wanpipeX.conf: TDMV_DCHAN = 24 zaptel.conf: hardhdlc = 24 (or in our case 48 since it's the second span) which means use wanpipe to handle the d-channel hdlc assuming the wanpipe driver has the necessary support (wanpipe on my platform doesn't). > Also, I recommend changing this line: > wbg1 = wanpipe2, , TDM_VOICE, Comment > > To this: > wbg1 = wanpipe2, , TDM_VOICE_API, Comment The sangoma voice API interface isn't available on my platform and shouldn't be necessary when using zaptel. > assuming that this is what you want then you will need to use > ozmod_libpri because the default OpenZAP PRI stack does not > currently support being the network side. Are you sure? Openzap appears to contain implementations for both NT and TE. The configuration file supports specifying either user or network for the mode. Is the NT support currently nonfunctional? I had tried configuring the Cisco as the NT with similar results. > I don't see where timing is specified It's the same T1 which was being used for RBS between FreeSWITCH and the Cisco so that timing (etc) should be okay. No errors are showing up at the physical level and the Cisco reports Layer 1 as active. The trace on the Cisco seems to show Layer 2 coming up (timestamps 22:53:44.264 through 22:54:21.760), then there's a long pause during which no Receive Ready frames are received from FreeSWITCH. At this point the Cisco gets unhappy and marks Layer 2 as down. If nothing obvious comes to anyone's mind, then I'll simply need to trace through the FreeSWITCH ISDN code and see what's going on. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From mszlazak at aol.com Wed Apr 8 16:15:35 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 08 Apr 2009 19:15:35 -0400 Subject: [Freeswitch-users] Problem with originate in javascript. Message-ID: <8CB86B70E569454-8D4-125F@MBLK-M41.sysops.aol.com> I want to run a script with a scheduler but I'm having a problem with how to set up the originate in Javascript. The originate would go something like: originate {id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/12223334444 at 10.0.0.5:5061 GINO_ANS I can get this to work: session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061"); But I want to "drop" that into an extension that runs another script and can't get either of these to work: session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061 GINO_ANS"); session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); Also, will I have problems running the second script from the first script? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/b5ff2c31/attachment-0002.html From wiltingtree at gmail.com Wed Apr 8 18:59:46 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Wed, 8 Apr 2009 21:59:46 -0400 Subject: [Freeswitch-users] Two or more simultaneous calls not Message-ID: Traun, thanks again for your help. I followed your advice and I made some progress! I tested with the latest trunk version and also with 1.0.2, and both exhibited the same behavior. I then tried writing a test script in Lua, and it worked fine. So this meant the problem was in the Python module (I was sure it was some FS config issue). So I started playing with a small test Python script, and I narrowed the problem down to when I'm using the "read" function. Here is my test script: from freeswitch import * def handler(session, args): #answer the call session.answer(); #play a file session.streamFile("long_prompt.mp3") # Test 1 - FAILED! digits = session.read(5, 10, "long_prompt.mp3", 3000, "#") # Test 2 - WORKED OK! #session.getDigits(1,"#",7000) #session.streamFile("long_prompt.mp3") # TEST 3 - WORKED OK! #digits = session.playAndGetDigits(5, 10, 1, 60, "#","long_prompt.mp3", "", "") #hangup session.hangup(); When I uncomment the code under test 1 and I make two simultaneous calls, the initial prompt plays for both calls just fine. But then the second prompt only plays on one of the channels and the other one just has dead air. When the first channel finishes playing the prompt, then the second channel starts playing it. Then I re-comment test 1 and uncomment either test 2 or test 3, Both prompts play just fine for both channels. So I think there may be a bug in the read() function somewhere. I took a look at it, but it's way over my head. Thanks again, Adam >Message: 8 >Date: Thu, 9 Apr 2009 02:36:08 +0430 >From: Traun Leyden >Subject: Re: [Freeswitch-users] Two or more simultaneous calls not >To: freeswitch-users at lists.freeswitch.org >Message-ID: > >Content-Type: text/plain; charset="iso-8859-1" >Hi Adam, >I'm stumped .. I guess you could try the following: >* Try with the trunk version of freeswitch. I don't think it will matter, >but just in case >* Try to simulate the same test with a Lua script. Do you see the same >problem? >If those don't turn up anything, then the next logical step would be >to start adding printf() statements in the mod_python code and >find out where it is getting stuck. In particular around the parts where >it swaps the threadstate in and out. I might be able to create a patch >for you, but try those other tests first. >HTH, >Traun > >> >> Thanks for the response Traun. The version of Python is 2.4.3, and I >> didn't >> build it myself, I installed it with yum. >> The version of Red Hat is 4.1.2-41. >> "import threading" works fine, so I don't think it's a Python threading >> issue. >> The FreeSWITCH version I installed is the >> freeswitch-1.0.3.tar.gz< >> http://files.freeswitch.org/freeswitch-1.0.3.tar.gz> >> located >> at files.freeswitch.org. >> I didn't make any major changes to the configuration; I enabled Python and >> set-up the SIP profile, directory and dialplan. No other changes. >> Any other help would be appreciated, since I really don't know where to >> look. >> >> Thanks, >> Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/aa0f15c8/attachment-0002.html From zhaoxxqq at 163.com Wed Apr 8 19:02:05 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Thu, 9 Apr 2009 10:02:05 +0800 Subject: [Freeswitch-users] Polycom register problem in private address Message-ID: <200904091002045745669@163.com> hi, I use FS server at public Address. I use polycom's IP550 at private address(192.168.0.120), Now there is a problem that the IP550 can not register to FS. But when I use account to eyebeam, the registering is OK. the attachment is my IP 550's config file, I think it must be NAT problem. Can anyone can help me solve it? 2009-04-09 zhaoxxqq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/c0b231a2/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: sip.cfg Type: application/octet-stream Size: 183561 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/c0b231a2/attachment-0004.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: phone[0004f2166b56].cfg Type: application/octet-stream Size: 12396 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/c0b231a2/attachment-0005.obj From brian at freeswitch.org Wed Apr 8 20:14:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Apr 2009 22:14:50 -0500 Subject: [Freeswitch-users] Polycom register problem in private address In-Reply-To: <200904091002045745669@163.com> References: <200904091002045745669@163.com> Message-ID: <819E811F-6CC6-407E-B689-2286926CD31D@freeswitch.org> This is because the Polycom doesn't support STUN, RPORT or any other nat traversal technology. You have a couple of choices please review http://wiki.freeswitch.org/wiki/NAT_Traversal Also review the NDLB-force-rport option for the sofia profile to assume rport. CAUTION this breaks things like cisco phones. /b On Apr 8, 2009, at 9:02 PM, zhaoxxqq wrote: > hi, > I use FS server at public Address. I use polycom's IP550 at private > address(192.168.0.120), Now there is a problem that the IP550 can > not register to FS. But when I use account to eyebeam, the > registering is OK. the attachment is my IP 550's config file, I > think it must be NAT problem. Can anyone can help me solve it? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090408/fec7b13b/attachment-0002.html From solko at gcdf.pl Thu Apr 9 00:51:03 2009 From: solko at gcdf.pl (Szymon Olko) Date: Thu, 09 Apr 2009 09:51:03 +0200 Subject: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption In-Reply-To: <8CB868AC2870B10-8D4-75@MBLK-M41.sysops.aol.com> References: <49DB8B14.70400@gmx.net><3072928E-D20C-48FA-B08B-1854C0548379@freeswitch.org> <49DBD946.5060406@gmx.net><8CB8602216B3FB4-14E8-2424@FWM-D28.sysops.aol.com> <49DC3BFA.4060200@gcdf.pl><8CB862C76FD6320-1E54-F7F@WEBMAIL-DG10.sim.aol.com> <49DC5953.8010000@gcdf.pl><8CB867CC493BA61-BA0-27A4@MBLK-M41.sysops.aol.com> <49DCE0A5.6080905@gcdf.pl> <8CB868AC2870B10-8D4-75@MBLK-M41.sysops.aol.com> Message-ID: <49DDA8E7.9060505@gcdf.pl> mszlazak at aol.com pisze: > Na jaki? czas, b?d? uczy? si?, jak radzi? sobie z Tortoise SVN b??d?w. > > M?j polski nie jest zbyt dobre, ale dzi?kuj?. Google pomaga. > Your polish is much better then my english. I think on that list we should stay with english. Where are you from? > -----Original Message----- > From: Szymon Olko > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, 8 Apr 2009 10:36 am > Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > 127.0.0.1 and memory consumption > > mszlazak at aol.com pisze: > >> OK , you're SVN updating on a Linux system but I'm using Windows. The > >> very few times I tried with Tortoise SVN I ran into problems were it > >> would fail because of some path not being present or some strange symbol > >> in a file or something else. Since I'm not experienced enough and don't > >> always have the time, I gave up on this approach and just start over > >> again in a different folder then reconfigure the updated FS and transfer > >> files from an older FS. Yup, it sucks. > >> > > > > Yes I'm linux user. > > If you have problems with svn update then you can do your way, make fresh > > checkout every time. After CO copy modules.conf and > > build new version, just copy old config files > to installation directory if it is > > always different one. > > > > That's why I hate gui tools for things like full svn update. > >> > >> > >> -----Original Message----- > >> From: Szymon Olko > > >> To: freeswitch-users at lists.freeswitch.org > >> Sent: Wed, 8 Apr 2009 12:59 am > >> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> 127.0.0.1 and memory consumption > >> > >> mszlazak at aol.com > pisze: > >> > >>> I'm not quite sure what your asking. > >> > >>> Are you saying that I could run the latest FS svn but in a way that uses > >> > >>> my "older" configuration files? If so then I don't, and don't know how > >> > >>> ... blush blush. > >> > >>> If that's the easiest thing to do then please tell me how. > >> > >>> Thanks. Mark. > >> > >> > >> > >> Exactly, I do it that way. > >> > >> For first time I gave installation prefix when configuring FS. You can stay > > with > >> > >> /usr/local/freeswich/. > >> > >> Now every time i call 'make current' and it does not overwrite my > > configuration > >> > >> file. > >> > >> > >> > >> In case of huge changes in modules I copy/merge my config file with the one > > from > >>=2 > 0 > >> svn. I did not have problems with it, because > >> > >> developers makes good default values for new configuration options. > >> > >> > >> > >> I don't know which modules do you use, but in ones I use configuration is not > >> > >> changes a lot, there are new options added which > >> > >> does not break old one. > >> > >> > >> > >> make install do not copy configuration files for me if they are already > >> > >> installed, I have that on production server and all test > >> > >> servers. I assume this is correct behavior and I'm not the only one work like > >> > >> that. > >> > >> I looked in Makefile and it tests for config file before installing, so it > > does > >> > >> not overwrite them. > >> > >> > >> > >> Regards > >> > >> Szymon > >> > >>> > >> > >>> -----Original Message----- > >> > >>> From: Szymon Olko >> > >> > >>> To: freeswitch-users at lists.freeswitch.org > > >> > >>> Sent: Tue, 7 Apr 2009 10:54 pm > >> > >>> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> > >>> 127.0.0.1 and memory consumption > >> > >>>20 > >> > >>> mszlazak at aol.com > > > ?>> pisze: > >> > >>> > >> > >>>> I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a > >> > >>> > >> > >>>> new trunk and I have to go through vars.xml, etc changing > >> > >>> > >> > >>>> $${local_ip_v4} like you did. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Is there a way to change $${local_ip_v4} in one place. That way one > >> > >>> > >> > >>>> wouldn't have remember all the locations that it needs to be changed? > >> > >>> > >> > >>>> > >> > >>> > >> > >>> My configuration is not updated when I compile new version and install it. Do > > > >> > >>> > >> > >>> you run FS with configuration path pointed to svn > >> > >>> > >> > >>> work dir? > >> > >>> > >> > >>> > >> > >>> > >> > >>>> -----Original Message----- > >> > >>> > >> > >>>> From: Peter P GMX > > > ?>>> > >> > >>> > >> > >>>> To: freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>>> Sent: Tue, 7 Apr 2009 3:52 pm > >> > >>> > >> > >>>> Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on > >> > >>> > >> > >>>> 127.0.0.1 and memory consumption > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Thanks Brian, > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> what I was actually looking for was to use a standard SIP soft phone > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> with some additional features. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> I finally manged to make FS li > sten on 127.0.0.1 the following way: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> vars.xml > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> internal.xml > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> The rest is standard configuration. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Now communication Laptop-internal is UDP on port 5060 and external via > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> TLS on p > ort 5081, so I have no open port 5060 to the internet. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Best regards > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Peter > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Brian West schrieb: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote: > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> 1st Question: Is that possible or is another solution preferrable? > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > ;> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> Just use FreeSWITCH with mod_portaudio. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> 2nd Question: How can I change the amount of memory FS tries to reserve > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> to an absolute minumum (I only have 1 call at a time). Currently it > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> tries to reserve about 360M if I read that right. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> Thats virtual. Look at RES. > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> Best regards > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>>> Peter > >> > >>> > >> > >>>> > >> > >>> > >> > >>> >>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> Brian West > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> brian at freeswitch.org > > > ?>> > > ?> > >> > >> ?>?>> > > ?> > >> > >> ?>?> > >> > >>> > >> > >>> ?> > > ?>?>?>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > ; > >>>>> -- Meet us a ClueCon! http://www.cluecon.com > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> ------------------------------------------------------------------------ > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> _______________________________________________ > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> Freeswitch-users mailing list > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> Freeswitch-users at lists.freeswitch.org > > > ?>> > > > >> > >> ?> > > ?>?>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> http://www.freeswitch.org > >> > >>> > >> > >>>> > >> > >>> > >> > >>>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> _______________________________________________ > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> Freeswitch-users mailing list > >> > >>> > >> > >>>> > >> > >>> > >> > >>> >> Freeswitch-users at lists.freeswitch.org > > > ?>> > > > >> > >> ?> > > ?>?>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> http://www.freeswitch.org > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> --------------------- > --------------------------------------------------- > >> > >>> > >> > >>>> *The Average US Credit Score is 692. See Yours in Just 2 Easy Steps! > >> > >>> > >> > >>>> * > > > > > >> > >> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> > >> > >>> > >> > >>>> ------------------------------------------------------------------------ > >> > >>> > >> > >>>> > >> > >>> > > & > gt; > >>>> _______________________________________________ > >> > >>> > >> > >>>> Freeswitch-users mailing list > >> > >>> > >> > >>>> Freeswitch-users at lists.freeswitch.org > > > ?>> > >> > >>> > >> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>>> http://www.freeswitch.org > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> _______________________________________________ > >> > >>> > >> > >>> Freeswitch-users mailing list > >> > >>> > >> > >>> Freeswitch-users at lists.freeswitch.org > > > ers at lists.freeswitch.org ?>> > >> > >>> > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> > >> > >>> http://www.freeswitch.org > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> New Deals on Dell Netbooks - Now starting at $299 > >> > >>> > >> > >>> > >> > >>> > >> > >>> > >> > >>> ------------------------------------------------------------------------ > >> > >>> > >> > >>> _______________________________________________ > >> > >>> Freeswitch-users mailing list > >> > >>> Freeswitch-users at lists.freeswitch.org > > >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >>> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> _______________________________________________ > >> > >> Freeswitch-users mailing list > >> > >> Freeswitch-users at lists.freeswitch.org > > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> ------------------------------------------------------------------------ > >> New Deals on Dell Netbooks - Now starting at $299 > >> > >> > >> > >> > >> --------------------------------------- > --------------------------------- > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 bmsword at gmail.com Thu Apr 9 01:13:10 2009 From: bmsword at gmail.com (bmsword) Date: Thu, 9 Apr 2009 16:13:10 +0800 Subject: [Freeswitch-users] Freeswitch as a media server Message-ID: <200904091613018903127@gmail.com> hi,all Can freeswitch be integrated with another softswitch as a media server? if it can, how to configure? thanks! andy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/f9f64ee4/attachment-0002.html From mchlmll at gmail.com Thu Apr 9 01:18:34 2009 From: mchlmll at gmail.com (Michele M) Date: Thu, 9 Apr 2009 10:18:34 +0200 Subject: [Freeswitch-users] How to design my project ? In-Reply-To: <7b197bef0904080956h190a0bb1v6fda1c6965931390@mail.gmail.com> References: <7b197bef0904080956h190a0bb1v6fda1c6965931390@mail.gmail.com> Message-ID: Ciao Giovanni, thanks for the quick answer, but still I can't get if it is possible to use the libfreeswitch w/o running FS to accomplish what I meant. It would need alot of time of development to make my application run as a FS application.Much better would be using libfreeswitch inside my application.But still the question arises:" Is Libfreeswitch enough for having sip endpoints and ASR/TTS API?" Do you have some examples for it? Thanks again Michele 2009/4/8 Giovanni Maruzzelli > Ciao Michele, > > as a start is definitely better (and more gratifying) that you runs > FreeSWITCH. > > Then, if (and only if) there is a compelling reason that justify the > amount of time needed to develop a standalone application, go for it. > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Apr 8, 2009 at 12:34 PM, Michele M wrote: > > Hi there, > > > > I'm quite a newbie about freeswitch. I have an application (IVR) that > > needs to have endpoints SIP to register,answer the calls and transfer > them > > to the right phones.(I( have my own SIP server).Moreover it needs also a > > ASR/TTS API' set to communicate with my ASR/TTS engine ( just for > example > > let's assume it is Cepstral). I'd wouldn't want to have freeswitch > running > > and communicate with it to accomplish that but just to use the > libfreeswitch > > library embedded. As I don't know that much about freeswitch can it be > done? > > or just I need to have freeswitch running as a must? Can somebody point > me > > to the right place where to find example of using library embedded (best > > examples for what I'm trying to do) as I have not found that many? > > > > Thanks in advance > > > > Miki > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/3a710d7f/attachment-0002.html From saigop at gmail.com Thu Apr 9 03:03:40 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Thu, 9 Apr 2009 15:33:40 +0530 Subject: [Freeswitch-users] Freeswitch as a media server In-Reply-To: <200904091613018903127@gmail.com> References: <200904091613018903127@gmail.com> Message-ID: <2ea4d47e0904090303q5006fb37le2b2fc97419b4c41@mail.gmail.com> Yes, you can connect freeswitch with another media gateway like audiocode or any softswitch. you can find here to connect with audiocode http://wiki.freeswitch.org/index.php?title=Configuring_AudioCodes_MP-114/118&printable=yes this is for analog audiocode, you can also connect with same setting with digital audiocodes. On Thu, Apr 9, 2009 at 1:43 PM, bmsword wrote: > hi,all > Can freeswitch be integrated with another softswitch as a media server? > if it can, how to configure? > thanks! > andy. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/479285f2/attachment-0002.html From ceino.no at gmail.com Thu Apr 9 03:28:09 2009 From: ceino.no at gmail.com (Ceino) Date: Thu, 09 Apr 2009 12:28:09 +0200 Subject: [Freeswitch-users] How to design my project ? In-Reply-To: References: <7b197bef0904080956h190a0bb1v6fda1c6965931390@mail.gmail.com> Message-ID: <49DDCDB9.1080509@gmail.com> Hi Miki, I'm not an expert on freeswitch but I'm sure that libfreeswitch can be embedded into c-applications (http://wiki.freeswitch.org/wiki/Embedding_FreeSWITCH). In addition it possible to write part of an application as an loadable module (applications in freeswitch are loadable modules). The reason for this is that a loadable module can use the dialplan framework (for example an incomping call can be routed to your application). It is also a good idea to look into how the scripting languages are used in FS (http://wiki.freeswitch.org/wiki/Languages_for_Call_Control). //Accordingly to voipinfo.org (http://www.voipinfo.org/wiki/view/FreeSwitch) is FS a library which ships with a small executable that loads the library, launches the core, and performs the various tasks that are defined by the modules (freeswitch applications). libfreeswitch includes sofia sip endpoint and ASR/TSS (native or through MRCP). You need to modify some configuration files to enable ASR/TSS (sip endpoint sofia is default). Best Regards Lars Sivertsen Michele M wrote: > Ciao Giovanni, > > thanks for the quick answer, but still I can't get if it is possible > to use the libfreeswitch w/o running FS to accomplish what I meant. > It would need alot of time of development to make my application run > as a FS application.Much better would be using libfreeswitch inside my > application.But still the question arises:" Is Libfreeswitch enough > for having sip endpoints and ASR/TTS API?" > Do you have some examples for it? > > Thanks again > > Michele > > 2009/4/8 Giovanni Maruzzelli > > > Ciao Michele, > > as a start is definitely better (and more gratifying) that you > runs FreeSWITCH. > > Then, if (and only if) there is a compelling reason that justify the > amount of time needed to develop a standalone application, go for it. > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Apr 8, 2009 at 12:34 PM, Michele M > wrote: > > Hi there, > > > > I'm quite a newbie about freeswitch. I have an application > (IVR) that > > needs to have endpoints SIP to register,answer the calls and > transfer them > > to the right phones.(I( have my own SIP server).Moreover it > needs also a > > ASR/TTS API' set to communicate with my ASR/TTS engine ( just > for example > > let's assume it is Cepstral). I'd wouldn't want to have > freeswitch running > > and communicate with it to accomplish that but just to use the > libfreeswitch > > library embedded. As I don't know that much about freeswitch can > it be done? > > or just I need to have freeswitch running as a must? Can > somebody point me > > to the right place where to find example of using library > embedded (best > > examples for what I'm trying to do) as I have not found that many? > > > > Thanks in advance > > > > Miki > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 9 06:09:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Apr 2009 08:09:27 -0500 Subject: [Freeswitch-users] Problem with originate in javascript. In-Reply-To: <8CB86B70E569454-8D4-125F@MBLK-M41.sysops.aol.com> References: <8CB86B70E569454-8D4-125F@MBLK-M41.sysops.aol.com> Message-ID: <191c3a030904090609n440e7a4n80444bcf7902113a@mail.gmail.com> The 2nd 2 examples you provided are invalid, they depict the usage of the originate api command in the context of the constructor to a JS session. If you want to send the call to another extension you have to create the channel like you did in the first example followed by session.execute("transfer", "GINO_ANS XML default"); at which time it would be wise if you deref the session object because its thread will be running in the new extension. A better way would be to do both in one with a single call to the originate api command apiExecute("originate", "{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ 14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); This never gives you a session object it just creates a channel and transfers it to the desired extension. A Documentation Re-factorial Engineer may be able to add it to the relevant page on the wiki if it is not already present. On Wed, Apr 8, 2009 at 6:15 PM, wrote: > I want to run a script with a scheduler but I'm having a problem with how > to set up the originate in Javascript. > > The originate would go something like: > > originate > {id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ > 12223334444 at 10.0.0.5:5061 GINO_ANS > > I can get this to work: > > session = new > Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ > 14082031170 at 10.0.0.5:5061"); > > But I want to "drop" that into an extension that runs another script and > can't get either of these to work: > > session = new > Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ > 14082031170 at 10.0.0.5:5061 GINO_ANS"); > > session = new > Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ > 14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); > > Also, will I have problems running the second script from the first script? > > Thanks. > > > > ------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/d2ce6528/attachment-0002.html From anthony.minessale at gmail.com Thu Apr 9 06:15:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Apr 2009 08:15:38 -0500 Subject: [Freeswitch-users] Two or more simultaneous calls not In-Reply-To: References: Message-ID: <191c3a030904090615j2b736b7bp890c903c00a014f6@mail.gmail.com> easier than you think, Just like Traun had suspected, the methods in the C++ wrapper for the read app was missing the begin and end allow threads calls that are only important to python to tell it to suspend the thread state while FS specific code is being executed. 2 line fix in trunk rev 12958 On Wed, Apr 8, 2009 at 8:59 PM, Adam Wilt wrote: > Traun, thanks again for your help. > I followed your advice and I made some progress! > > I tested with the latest trunk version and also with 1.0.2, and both > exhibited the same behavior. > I then tried writing a test script in Lua, and it worked fine. > So this meant the problem was in the Python module (I was sure it was some > FS config issue). > So I started playing with a small test Python script, and I narrowed the > problem down to when I'm using the "read" function. > Here is my test script: > > from freeswitch import * > def handler(session, args): > #answer the call > session.answer(); > #play a file > session.streamFile("long_prompt.mp3") > # Test 1 - FAILED! > digits = session.read(5, 10, "long_prompt.mp3", 3000, "#") > # Test 2 - WORKED OK! > #session.getDigits(1,"#",7000) > #session.streamFile("long_prompt.mp3") > # TEST 3 - WORKED OK! > #digits = session.playAndGetDigits(5, 10, 1, 60, "#","long_prompt.mp3", > "", "") > #hangup > session.hangup(); > When I uncomment the code under test 1 and I make two simultaneous calls, > the initial prompt plays for both calls just fine. But then the second > prompt only plays on one of the channels and the other one just has dead > air. When the first channel finishes playing the prompt, then the second > channel starts playing it. > > Then I re-comment test 1 and uncomment either test 2 or test 3, Both > prompts play just fine for both channels. > > So I think there may be a bug in the read() function somewhere. I took a > look at it, but it's way over my head. > > Thanks again, > Adam > > > > >Message: 8 > >Date: Thu, 9 Apr 2009 02:36:08 +0430 > >From: Traun Leyden > >Subject: Re: [Freeswitch-users] Two or more simultaneous calls not > >To: freeswitch-users at lists.freeswitch.org > >Message-ID: > > > >Content-Type: text/plain; charset="iso-8859-1" > > >Hi Adam, > > >I'm stumped .. I guess you could try the following: > > >* Try with the trunk version of freeswitch. I don't think it will matter, > >but just in case > > >* Try to simulate the same test with a Lua script. Do you see the same > >problem? > > >If those don't turn up anything, then the next logical step would be > >to start adding printf() statements in the mod_python code and > >find out where it is getting stuck. In particular around the parts where > >it swaps the threadstate in and out. I might be able to create a patch > >for you, but try those other tests first. > > >HTH, > >Traun > > > > >> > >> Thanks for the response Traun. The version of Python is 2.4.3, and I > >> didn't > >> build it myself, I installed it with yum. > >> The version of Red Hat is 4.1.2-41. > >> "import threading" works fine, so I don't think it's a Python threading > >> issue. > >> The FreeSWITCH version I installed is the > >> freeswitch-1.0.3.tar.gz< > >> http://files.freeswitch.org/freeswitch-1.0.3.tar.gz> > >> located > >> at files.freeswitch.org. > >> I didn't make any major changes to the configuration; I enabled Python > and > >> set-up the SIP profile, directory and dialplan. No other changes. > >> Any other help would be appreciated, since I really don't know where to > >> look. > >> > >> Thanks, > >> 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/014c661b/attachment-0002.html From peter.olsson at visionutveckling.se Thu Apr 9 06:34:09 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 9 Apr 2009 15:34:09 +0200 Subject: [Freeswitch-users] Crash in mod_opal when hanging up call... Message-ID: Hello everyone. I've been following this project for quite some time now, but I never got the time to test it. But today I finally had a day off from work, so I could sit and play around with it for some time :) Everything wen't really smooth - even though I built everything from scratch on a Windows machine, including mod_opal (linked against the opal library). And with the docs I found I didn't even have to search Google to get it up and running :) So first of all - what a great job, guys - I'm really impressed, and the code seems really stable as well! My setup right now is FreeSWITCH in Windows XP, with both SIP and H323 enabled. And most of the stuff works just fine. However, I think I've found a bug in mod_opal - it sometimes causes FreeSWITCH to crash when hanging up a call. I think that mod_opal is considered to be in beta stage still, so I'm not all that surprised. :) Check the error found in the log below. Does anyone have any ideas? It's pretty easy to reproduce, just dial in to FreeSWITCH using H323 and hang up the call, for me it happens maybe 2 out of 5 times. I'm using latest SVN trunk versions (checked out today), for both FreeSWITCH and for opal/ptlib. If you need further information, or if I should file a jira case, please get back to me, and I'll try to help out as much as possible. 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:657 FSConnection::OnAlerting() Ring-Ready ! 2009-04-09 14:37:05 [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel opal/in:9999 [c7441e16-394c-d843-9ce4-760786dcecbf] 2009-04-09 14:37:05 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Peter Olsson [172.18.96.100]->9999 in context default 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 initialise opal/in:9999write audio codec G.711-uLaw-64k for connection FSMediaStream-Source-G.711-uLaw-64k 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 initialise opal/in:9999read audio codec G.711-uLaw-64k for connection FSMediaStream-Sink-G.711-uLaw-64k 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:795 FSConnection::OnOpenMediaStream() Channel [opal/in:9999] has been answered Assertion failed: (*frame)->codec != ((void *)0), file ..\..\src\switch_core_io.c, line 202 2009-04-09 14:37:09 [NOTICE] mod_opal.cpp:650 FSConnection::OnReleased() Hangup opal/in:9999 [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-09 14:37:09 [INFO] h323pdu.cxx:1005 H225() Read error (0): From peter.olsson at visionutveckling.se Thu Apr 9 06:37:17 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 9 Apr 2009 15:37:17 +0200 Subject: [Freeswitch-users] Status of Sangoma support in Windows? Message-ID: >From what I've found in the docs and lists, the support for Sangoma (PRI) cards is still not avaiable in the Windows port. Is this planned to be implemented in the future, or will it never be included in Win32? Just a curious thought, since I might need to use some PRI stuff in the future... Regards, Peter Olsson From mike at jerris.com Thu Apr 9 07:05:32 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 9 Apr 2009 10:05:32 -0400 Subject: [Freeswitch-users] Status of Sangoma support in Windows? In-Reply-To: References: Message-ID: <243EF9A5-48F0-45CE-BFB6-8CB54040679B@jerris.com> The code that "should" work for this is on a box at Sangmoa under testing right now on linux. It should be committed as soon as the new driver is released (which the new module will require) at which point it will just need build integration completed and proper testing on windows. Mike On Apr 9, 2009, at 9:37 AM, Peter Olsson wrote: >> From what I've found in the docs and lists, the support for Sangoma >> (PRI) cards is still not avaiable in the Windows port. Is this >> planned to be implemented in the future, or will it never be >> included in Win32? Just a curious thought, since I might need to >> use some PRI stuff in the future... > > Regards, > > Peter Olsson From peter.olsson at visionutveckling.se Thu Apr 9 07:31:29 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 9 Apr 2009 16:31:29 +0200 Subject: [Freeswitch-users] Status of Sangoma support in Windows? In-Reply-To: <243EF9A5-48F0-45CE-BFB6-8CB54040679B@jerris.com> Message-ID: That sounds great - we'll just have to hope for the best :) Thanks for your quick response. //Peter On 09-04-09 16.05, "Michael Jerris" wrote: The code that "should" work for this is on a box at Sangmoa under testing right now on linux. It should be committed as soon as the new driver is released (which the new module will require) at which point it will just need build integration completed and proper testing on windows. Mike On Apr 9, 2009, at 9:37 AM, Peter Olsson wrote: >> From what I've found in the docs and lists, the support for Sangoma >> (PRI) cards is still not avaiable in the Windows port. Is this >> planned to be implemented in the future, or will it never be >> included in Win32? Just a curious thought, since I might need to >> use some PRI stuff in the future... > > Regards, > > Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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:49de027e32935992579123! From peder at networkoblivion.com Thu Apr 9 07:59:56 2009 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Thu, 09 Apr 2009 09:59:56 -0500 Subject: [Freeswitch-users] Polycom register problem in private address In-Reply-To: <819E811F-6CC6-407E-B689-2286926CD31D@freeswitch.org> References: <200904091002045745669@163.com> <819E811F-6CC6-407E-B689-2286926CD31D@freeswitch.org> Message-ID: <49DE0D6C.1080503@networkoblivion.com> You might try entering your external NAT IP into the Polycom config. I've found that if you specify the external IP, Polycom's generally work better thru NAT. This is one area where Cisco is superior to Polycom. On Cisco, you just enable NAT and you don't have to specify the external IP. Of course Cisco has a whole mess of their own issues too. Peder Brian West wrote: > This is because the Polycom doesn't support STUN, RPORT or any other nat > traversal technology. You have a couple of choices please > review http://wiki.freeswitch.org/wiki/NAT_Traversal > > Also review the NDLB-force-rport option for the sofia profile to assume > rport. CAUTION this breaks things like cisco phones. > > /b > > On Apr 8, 2009, at 9:02 PM, zhaoxxqq wrote: > >> hi, >> I use FS server at public Address. I use polycom's IP550 at private >> address(192.168.0.120), Now there is a problem that the IP550 can not >> register to FS. But when I use account to eyebeam, the registering is >> OK. the attachment is my IP 550's config file, I think it must be NAT >> problem. Can anyone can help me solve it? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 9 08:38:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Apr 2009 08:38:25 -0700 Subject: [Freeswitch-users] need help getting ISDN talking to Cisco 3845 In-Reply-To: <200904082219.n38MJ0ld006139@jwlab.FEITH.COM> References: <200904082219.n38MJ0ld006139@jwlab.FEITH.COM> Message-ID: <87f2f3b90904090838i40bf4160l1e4f1bc51f8129be@mail.gmail.com> John, Just curious - why are you using zaptel at all? Does it provide something for you that the wanpipe drivers do not? I use Sangoma only with Sangoma cards and I have a lot of success. -MC On Wed, Apr 8, 2009 at 3:19 PM, John Wehle wrote: > > Okay, a few things. First off, the wanpipe2.conf file has a booboo. > > Don't think so. > > > This line is WRONG: > > TDMV_DCHAN = 0 > > Not exactly. My understanding is you can use either: > > wanpipeX.conf: TDMV_DCHAN = 0 > zaptel.conf: dchan = 24 (or in our case 48 since it's the second span) > > which means use zaptel to handle the d-channel hdlc or > > wanpipeX.conf: TDMV_DCHAN = 24 > zaptel.conf: hardhdlc = 24 (or in our case 48 since it's the second span) > > which means use wanpipe to handle the d-channel hdlc assuming the > wanpipe driver has the necessary support (wanpipe on my platform > doesn't). > > > Also, I recommend changing this line: > > wbg1 = wanpipe2, , TDM_VOICE, Comment > > > > To this: > > wbg1 = wanpipe2, , TDM_VOICE_API, Comment > > The sangoma voice API interface isn't available on my platform > and shouldn't be necessary when using zaptel. > > > assuming that this is what you want then you will need to use > > ozmod_libpri because the default OpenZAP PRI stack does not > > currently support being the network side. > > Are you sure? Openzap appears to contain implementations for > both NT and TE. The configuration file supports specifying > either user or network for the mode. Is the NT support > currently nonfunctional? > > I had tried configuring the Cisco as the NT with similar > results. > > > I don't see where timing is specified > > It's the same T1 which was being used for RBS between > FreeSWITCH and the Cisco so that timing (etc) should > be okay. No errors are showing up at the physical > level and the Cisco reports Layer 1 as active. > > The trace on the Cisco seems to show Layer 2 coming up > (timestamps 22:53:44.264 through 22:54:21.760), then > there's a long pause during which no Receive Ready > frames are received from FreeSWITCH. At this point > the Cisco gets unhappy and marks Layer 2 as down. > > If nothing obvious comes to anyone's mind, then I'll > simply need to trace through the FreeSWITCH ISDN code > and see what's going on. > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | > | John Wehle | Fax: 1-215-540-5495 | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/be59339c/attachment-0002.html From chris at fowler.cc Thu Apr 9 08:51:27 2009 From: chris at fowler.cc (Chris Fowler) Date: Thu, 09 Apr 2009 08:51:27 -0700 Subject: [Freeswitch-users] Polycom register problem in private address Message-ID: <1239292287.27625.1309790615@webmail.messagingengine.com> I'm running FS on Amazons' EC2 compute cloud (AWS) and have 30 Polycom phones working happily in this config. I modified the Internal profile in /usr/local/freeswitch/conf/sip_profiles/internal.xml to include: The phones connect on port 5060 - nothing specical to config in the -phone.cfg file for the phone; just host, port, user/pass. Chris. From brian at freeswitch.org Thu Apr 9 09:09:48 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Apr 2009 11:09:48 -0500 Subject: [Freeswitch-users] Polycom register problem in private address In-Reply-To: <1239292287.27625.1309790615@webmail.messagingengine.com> References: <1239292287.27625.1309790615@webmail.messagingengine.com> Message-ID: <45D90D9C-39D8-4B09-BFE2-77523151F13A@freeswitch.org> Did you request public IP's for your EC2 instance? /b On Apr 9, 2009, at 10:51 AM, Chris Fowler wrote: > I'm running FS on Amazons' EC2 compute cloud (AWS) and have 30 Polycom > phones working happily in this config. > > I modified the Internal profile in > /usr/local/freeswitch/conf/sip_profiles/internal.xml to include: > > > > > The phones connect on port 5060 - nothing specical to config in the > -phone.cfg file for the phone; just host, port, user/pass. > > Chris. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/35cd52f3/attachment-0002.html From msc at freeswitch.org Thu Apr 9 09:16:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Apr 2009 09:16:36 -0700 Subject: [Freeswitch-users] Polycom register problem in private address In-Reply-To: <1239292287.27625.1309790615@webmail.messagingengine.com> References: <1239292287.27625.1309790615@webmail.messagingengine.com> Message-ID: <87f2f3b90904090916g9926c5aj304035602e8c412b@mail.gmail.com> Hey, this would be great info to put on the wiki... (hint hint wink wink nudge nudge) :) -MC On Thu, Apr 9, 2009 at 8:51 AM, Chris Fowler wrote: > I'm running FS on Amazons' EC2 compute cloud (AWS) and have 30 Polycom > phones working happily in this config. > > I modified the Internal profile in > /usr/local/freeswitch/conf/sip_profiles/internal.xml to include: > > > > > The phones connect on port 5060 - nothing specical to config in the > -phone.cfg file for the phone; just host, port, user/pass. > > Chris. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/3e0bb54f/attachment-0002.html From mszlazak at aol.com Thu Apr 9 09:22:49 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Thu, 09 Apr 2009 12:22:49 -0400 Subject: [Freeswitch-users] Problem with originate in javascript. In-Reply-To: <191c3a030904090609n440e7a4n80444bcf7902113a@mail.gmail.com> References: <8CB86B70E569454-8D4-125F@MBLK-M41.sysops.aol.com> <191c3a030904090609n440e7a4n80444bcf7902113a@mail.gmail.com> Message-ID: <8CB87468E0848B5-874-626@WEBMAIL-MY06.sysops.aol.com> Hi Tony, But I thought we settled on "Janitor." ;-) BTW, it was the other point about keeping the FS founders involved or not in the documentation process that concerned much much more. That was the big issue that got me going on that thread. Anyway, I appreciate your help and will do some "document engineering" but need one further elaboration on de-referencing the original session object since I tried the execute("transfer" ...) before and couldn't get that to work. Can you show me an example and I can then put up both approaches. Mark. -----Original Message----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thu, 9 Apr 2009 6:09 am Subject: Re: [Freeswitch-users] Problem with originate in javascript. The 2nd 2 examples you provided are invalid, they depict the usage of the originate api command in the context of the constructor to a JS session. If you want to send the call to another extension you have to create the channel like you did in the first example followed by session.execute("transfer", "GINO_ANS XML default"); at which time it would be wise if you deref the session object because its thread will be running in the new extension. A better way would be to do both in one with a single call to the originate api command apiExecute("originate", "{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); This never gives you a session object it just creates a channel and transfers it to the desired extension. A Documentation Re-factorial Engineer may be able to add it to the relevant page on the wiki if it is not already present. On Wed, Apr 8, 2009 at 6:15 PM, wrote: I want to run a script with a scheduler but I'm having a problem with how to set up the originate in Javascript. The originate would go something like: originate {id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/12223334444 at 10.0.0.5:5061 GINO_ANS I can get this to work: session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061"); But I want to "drop" that into an extension that runs another script and can't get either of these to work: session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061 GINO_ANS"); session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); Also, will I have problems running the second script from the first script? Thanks. New Deals on Dell Netbooks - Now starting at $299 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/3a38fa31/attachment-0002.html From chris at fowler.cc Thu Apr 9 09:30:18 2009 From: chris at fowler.cc (Chris Fowler) Date: Thu, 09 Apr 2009 09:30:18 -0700 Subject: [Freeswitch-users] Polycom register problem in private address In-Reply-To: <45D90D9C-39D8-4B09-BFE2-77523151F13A@freeswitch.org> References: <1239292287.27625.1309790615@webmail.messagingengine.com> <45D90D9C-39D8-4B09-BFE2-77523151F13A@freeswitch.org> Message-ID: <1239294618.4932.1309797565@webmail.messagingengine.com> Brian: Did you request public IP's for your EC2 instance? Yes; there is an Elastic IP (EIP) associated with the instance. Also specify the EIP in vars.xml >> Re: Wiki Yup I need to get on this. FWIW - I work for RightScale; our computer room is empty except for routers and switches. *Everything* else lives in the Cloud :-) Cheers, Chris. From anthony.minessale at gmail.com Thu Apr 9 09:40:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Apr 2009 11:40:32 -0500 Subject: [Freeswitch-users] Problem with originate in javascript. In-Reply-To: <8CB87468E0848B5-874-626@WEBMAIL-MY06.sysops.aol.com> References: <8CB86B70E569454-8D4-125F@MBLK-M41.sysops.aol.com> <191c3a030904090609n440e7a4n80444bcf7902113a@mail.gmail.com> <8CB87468E0848B5-874-626@WEBMAIL-MY06.sysops.aol.com> Message-ID: <191c3a030904090940l6e6ad6c7wfb64cf6fd48afd30@mail.gmail.com> I think I forgot to mention you need to session.setAutoHangup(false) to stop the channel from being auto-hungup when it goes out of scope of the script. session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ 14082031170 at 10.0.0.5:5061"); session.setAutoHangup(false); session.execute("transfer", "GINO_ANS XML default"); session = undefined; On Thu, Apr 9, 2009 at 11:22 AM, wrote: > Hi Tony, > > But I thought we settled on "Janitor." ;-) > BTW, it was the other point about keeping the FS founders involved or not > in the documentation process that concerned much much more. That was the big > issue that got me going on that thread. > > Anyway, I appreciate your help and will do some "document engineering" but > need one further elaboration on de-referencing the original session object > since I tried the execute("transfer" ...) before and couldn't get that to > work. Can you show me an example and I can then put up both approaches. > > Mark. > > > -----Original Message----- > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, 9 Apr 2009 6:09 am > Subject: Re: [Freeswitch-users] Problem with originate in javascript. > > The 2nd 2 examples you provided are invalid, they depict the usage of the > originate api command in the context of the constructor > to a JS session. > > If you want to send the call to another extension you have to create the > channel like you did in the first example followed by > session.execute("transfer", "GINO_ANS XML default"); > at which time it would be wise if you deref the session object because its > thread will be running in the new extension. > > > A better way would be to do both in one with a single call to the originate > api command > > apiExecute("originate", > "{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ > 14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); > > This never gives you a session object it just creates a channel and > transfers it to the desired extension. > > > A Documentation Re-factorial Engineer may be able to add it to the relevant > page on the wiki if it is not already present. > > > > On Wed, Apr 8, 2009 at 6:15 PM, wrote: > >> I want to run a script with a scheduler but I'm having a problem with how >> to set up the originate in Javascript. >> >> The originate would go something like: >> >> originate >> {id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ >> 12223334444 at 10.0.0.5:5061 GINO_ANS >> >> I can get this to work: >> >> session = new >> Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ >> 14082031170 at 10.0.0.5:5061"); >> >> But I want to "drop" that into an extension that runs another script and >> can't get either of these to work: >> >> session = new >> Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ >> 14082031170 at 10.0.0.5:5061 GINO_ANS"); >> >> session = new >> Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/ >> 14082031170 at 10.0.0.5:5061 GINO_ANS XML default"); >> >> Also, will I have problems running the second script from the first >> script? >> >> Thanks. >> >> >> >> ------------------------------ >> New Deals on Dell Netbooks - Now starting at $299 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > New Deals on Dell Netbooks - Now starting at $299 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/82ead3fe/attachment-0002.html From anthony.minessale at gmail.com Thu Apr 9 15:03:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Apr 2009 17:03:05 -0500 Subject: [Freeswitch-users] Crash in mod_opal when hanging up call... In-Reply-To: References: Message-ID: <191c3a030904091503g69885e9aj191da55d1162dc62@mail.gmail.com> can you try the latest trunk again r12975 if it's still a problem please open a jira on the issue http://jira.freeswitch.org On Thu, Apr 9, 2009 at 8:34 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Hello everyone. I've been following this project for quite some time now, > but I never got the time to test it. But today I finally had a day off from > work, so I could sit and play around with it for some time :) Everything > wen't really smooth - even though I built everything from scratch on a > Windows machine, including mod_opal (linked against the opal library). And > with the docs I found I didn't even have to search Google to get it up and > running :) So first of all - what a great job, guys - I'm really impressed, > and the code seems really stable as well! > > My setup right now is FreeSWITCH in Windows XP, with both SIP and H323 > enabled. And most of the stuff works just fine. However, I think I've found > a bug in mod_opal - it sometimes causes FreeSWITCH to crash when hanging up > a call. I think that mod_opal is considered to be in beta stage still, so > I'm not all that surprised. :) Check the error found in the log below. Does > anyone have any ideas? It's pretty easy to reproduce, just dial in to > FreeSWITCH using H323 and hang up the call, for me it happens maybe 2 out of > 5 times. > > I'm using latest SVN trunk versions (checked out today), for both > FreeSWITCH and for opal/ptlib. > > If you need further information, or if I should file a jira case, please > get back to me, and I'll try to help out as much as possible. > > 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:657 FSConnection::OnAlerting() > Ring-Ready ! > 2009-04-09 14:37:05 [NOTICE] switch_channel.c:597 switch_channel_set_name() > New Channel opal/in:9999 [c7441e16-394c-d843-9ce4-760786dcecbf] > 2009-04-09 14:37:05 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing Peter Olsson [172.18.96.100]->9999 in context default > 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 > initialise opal/in:9999write audio codec G.711-uLaw-64k for connection > FSMediaStream-Source-G.711-uLaw-64k > 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 > initialise opal/in:9999read audio codec G.711-uLaw-64k for connection > FSMediaStream-Sink-G.711-uLaw-64k > 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:795 > FSConnection::OnOpenMediaStream() Channel [opal/in:9999] has been answered > Assertion failed: (*frame)->codec != ((void *)0), file > ..\..\src\switch_core_io.c, line 202 > 2009-04-09 14:37:09 [NOTICE] mod_opal.cpp:650 FSConnection::OnReleased() > Hangup opal/in:9999 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-04-09 14:37:09 [INFO] h323pdu.cxx:1005 H225() Read error (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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090409/ce9178e8/attachment-0002.html From tleyden at branchcut.com Thu Apr 9 22:35:51 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Fri, 10 Apr 2009 10:05:51 +0430 Subject: [Freeswitch-users] Two or more simultaneous calls not Message-ID: Hey you beat me to it. I was going to have a look this morning but had no internet because some asswipe cut a bunch of fiber optic cables and took out phone/internet for a big part of the bay area. I haven't tried your patch yet, but I see something that looks suspect: http://fisheye.freeswitch.org/browse/FreeSWITCH/src/switch_cpp.cpp?r=12958 In playAndGetDigits() there are now two calls to begin_allow_threads() (line 778 and 780) followed by only one call to end_allow_threads() (line 793) Also I guess it would have better to test against JS, since it should have had the same bug right? Lua just ignores the threadswapping stuff but IIRC javascript uses it in much the same way as python. Or did I miss something? > Message: 5 > Date: Thu, 9 Apr 2009 08:15:38 -0500 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Two or more simultaneous calls not > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030904090615j2b736b7bp890c903c00a014f6 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > easier than you think, > > Just like Traun had suspected, the methods in the C++ wrapper for the read > app was missing the begin and end allow threads calls > that are only important to python to tell it to suspend the thread state > while FS specific code is being executed. > > 2 line fix in trunk rev 12958 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090410/1dfcf048/attachment-0002.html From excelsio at gmx.net Fri Apr 10 00:50:42 2009 From: excelsio at gmx.net (excelsio at gmx.net) Date: Fri, 10 Apr 2009 09:50:42 +0200 Subject: [Freeswitch-users] encryption gateway/proxy with freeswitch? Message-ID: <20090410075042.94340@gmx.net> Hi, the Alcatel OmniPCX Enterprise (OXE) of one of our customers seems to support the following encyption scenarios: - IP-phone on Alcatel OXE <=SRTP=> IP-phone on Alcatel OXE - IP-phone <=SIPS/SRTP=> Alcatel OXE <=> landline phone Unfortunately the Alcatel OXE doesn?t support SIPS/SRTP encryption betwenn itself and a SIP provider. So, for now it looks like: - Alcatel OXE <=SIP/RTP=> SIP provider The goal is, to encrypt that traffic, too. The SIP provider does support SRTP. So I?m asking myself whether to place a freeswitch between both which proxies and also encrypts the traffic? - Alcatel OXE <=SIP/STP> Freeswitch <=(SIPS)/SRTP=> SIP provider Has someone done something like this already? thanks in advance Michael From peter.olsson at visionutveckling.se Fri Apr 10 01:43:48 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 10 Apr 2009 10:43:48 +0200 Subject: [Freeswitch-users] Crash in mod_opal when hanging up call... In-Reply-To: <191c3a030904091503g69885e9aj191da55d1162dc62@mail.gmail.com> Message-ID: Thanks for the reply. This update got rid of the error in the log, but FreeSWITCH still crashes. I've files a jira cace about it (MODOPAL-3). On 09-04-10 00.03, "Anthony Minessale" wrote: can you try the latest trunk again r12975 if it's still a problem please open a jira on the issue http://jira.freeswitch.org On Thu, Apr 9, 2009 at 8:34 AM, Peter Olsson wrote: Hello everyone. I've been following this project for quite some time now, but I never got the time to test it. But today I finally had a day off from work, so I could sit and play around with it for some time :) Everything wen't really smooth - even though I built everything from scratch on a Windows machine, including mod_opal (linked against the opal library). And with the docs I found I didn't even have to search Google to get it up and running :) So first of all - what a great job, guys - I'm really impressed, and the code seems really stable as well! My setup right now is FreeSWITCH in Windows XP, with both SIP and H323 enabled. And most of the stuff works just fine. However, I think I've found a bug in mod_opal - it sometimes causes FreeSWITCH to crash when hanging up a call. I think that mod_opal is considered to be in beta stage still, so I'm not all that surprised. :) Check the error found in the log below. Does anyone have any ideas? It's pretty easy to reproduce, just dial in to FreeSWITCH using H323 and hang up the call, for me it happens maybe 2 out of 5 times. I'm using latest SVN trunk versions (checked out today), for both FreeSWITCH and for opal/ptlib. If you need further information, or if I should file a jira case, please get back to me, and I'll try to help out as much as possible. 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:657 FSConnection::OnAlerting() Ring-Ready ! 2009-04-09 14:37:05 [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel opal/in:9999 [c7441e16-394c-d843-9ce4-760786dcecbf] 2009-04-09 14:37:05 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Peter Olsson [172.18.96.100]->9999 in context default 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 initialise opal/in:9999write audio codec G.711-uLaw-64k for connection FSMediaStream-Source-G.711-uLaw-64k 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 initialise opal/in:9999read audio codec G.711-uLaw-64k for connection FSMediaStream-Sink-G.711-uLaw-64k 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:795 FSConnection::OnOpenMediaStream() Channel [opal/in:9999] has been answered Assertion failed: (*frame)->codec != ((void *)0), file ..\..\src\switch_core_io.c, line 202 2009-04-09 14:37:09 [NOTICE] mod_opal.cpp:650 FSConnection::OnReleased() Hangup opal/in:9999 [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-09 14:37:09 [INFO] h323pdu.cxx:1005 H225() Read error (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 From peter.olsson at visionutveckling.se Fri Apr 10 01:46:54 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 10 Apr 2009 10:46:54 +0200 Subject: [Freeswitch-users] mod_opal and DTMF... Message-ID: When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem to detect DTMF. I'm not sure if this is a setting somewhere in the config files, or if it's a bug. The test scenario is simple - use the default FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't detect any DTMF tones. Regards, Peter Olsson From peter.olsson at visionutveckling.se Fri Apr 10 01:59:25 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 10 Apr 2009 10:59:25 +0200 Subject: [Freeswitch-users] Some spidermonkey modules fails to load in Windows Message-ID: The spidermonkey modules core_db/odbc, curl, socket and teletone fails to load in Windows. They just return error 1271 (Sym Error). I'm not sure if this is a known issue, or if just doesn't work in Windows :) I've been using the latest SVN when trying this. Any ideas anyone? :) Regards, Peter Olsson From peter.olsson at visionutveckling.se Fri Apr 10 03:05:06 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 10 Apr 2009 12:05:06 +0200 Subject: [Freeswitch-users] How to find/build FreeSWITCH.Managed.dll? Message-ID: Hello again! When trying to load the mod_managed module it get an error that it can't find FreeSWITCH.Managed.dll. So my question is simply - where do I find this file, or how do I build it? I'm using the VC++ Express edition when building, so I guess I also have to install the C# edition - will this solve my problem? Sorry for asking stupid questions here - but I've just been playing around with FreeSWITCH for a day or so :) Regards, Peter Olsson From mike at jerris.com Fri Apr 10 05:19:18 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Apr 2009 08:19:18 -0400 Subject: [Freeswitch-users] Two or more simultaneous calls not In-Reply-To: References: Message-ID: <2DBE8205-E99A-4387-A727-B922B44A3236@jerris.com> On Apr 10, 2009, at 1:35 AM, Traun Leyden wrote: > > Hey you beat me to it. I was going to have a look this morning but > had no internet because some asswipe cut a bunch of fiber optic cables > and took out phone/internet for a big part of the bay area. > > I haven't tried your patch yet, but I see something that looks > suspect: > > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/switch_cpp.cpp?r=12958 > > In playAndGetDigits() there are now two calls to begin_allow_threads() > (line 778 and 780) followed by only one call to end_allow_threads() > (line 793) > > Also I guess it would have better to test against JS, since it > should have > had the same bug right? Lua just ignores the threadswapping stuff but > IIRC javascript uses it in much the same way as python. Or did I > miss something? > It's similar, but we don't use swig for javascript and we don't use switch_cpp so it would not have reproduced the issue. MIke -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090410/37d3794b/attachment-0002.html From mike at jerris.com Fri Apr 10 05:26:29 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Apr 2009 08:26:29 -0400 Subject: [Freeswitch-users] Two or more simultaneous calls not In-Reply-To: References: Message-ID: On Apr 10, 2009, at 1:35 AM, Traun Leyden wrote: > > Hey you beat me to it. I was going to have a look this morning but > had no internet because some asswipe cut a bunch of fiber optic cables > and took out phone/internet for a big part of the bay area. > > I haven't tried your patch yet, but I see something that looks > suspect: > > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/switch_cpp.cpp?r=12958 > > In playAndGetDigits() there are now two calls to begin_allow_threads() > (line 778 and 780) followed by only one call to end_allow_threads() > (line 793) Extra line removed, I did notice when looking in that file other methods that probably need begin/end, for example the blocking pop in event consumer. Would you mind going through the rest and seeing if their are other obvious misses? Mike > > Also I guess it would have better to test against JS, since it > should have > had the same bug right? Lua just ignores the threadswapping stuff but > IIRC javascript uses it in much the same way as python. Or did I > miss something? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090410/f276344b/attachment-0002.html From mike at jerris.com Fri Apr 10 05:29:18 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Apr 2009 08:29:18 -0400 Subject: [Freeswitch-users] Some spidermonkey modules fails to load in Windows In-Reply-To: References: Message-ID: <115D04CB-3FD5-4018-8667-F5F45AD405E6@jerris.com> On Apr 10, 2009, at 4:59 AM, Peter Olsson wrote: > The spidermonkey modules core_db/odbc, curl, socket and teletone > fails to load in Windows. They just return error 1271 (Sym Error). > I'm not sure if this is a known issue, or if just doesn't work in > Windows :) I've been using the latest SVN when trying this. > > Any ideas anyone? :) This was fixed in svn yesterday. Mike From mike at jerris.com Fri Apr 10 05:33:14 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Apr 2009 08:33:14 -0400 Subject: [Freeswitch-users] How to find/build FreeSWITCH.Managed.dll? In-Reply-To: References: Message-ID: <04C59CF7-6285-4CCD-818A-B71360F91B8E@jerris.com> On Apr 10, 2009, at 6:05 AM, Peter Olsson wrote: > When trying to load the mod_managed module it get an error that it > can't find FreeSWITCH.Managed.dll. > > So my question is simply - where do I find this file, or how do I > build it? I'm using the VC++ Express edition when building, so I > guess I also have to install the C# edition - will this solve my > problem? > > Sorry for asking stupid questions here - but I've just been playing > around with FreeSWITCH for a day or so :) Unfortunately Express edition does not allow for mixed soulutions so you need to go build the managed dll manually. The file is in src/mod/ languages/mod_managed/managed. Mike From peter.olsson at visionutveckling.se Fri Apr 10 06:11:35 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 10 Apr 2009 15:11:35 +0200 Subject: [Freeswitch-users] Some spidermonkey modules fails to load in Windows In-Reply-To: <115D04CB-3FD5-4018-8667-F5F45AD405E6@jerris.com> Message-ID: I looked into the SVN logs early this morning, and found out that something was changed for this. However, the problem still exists for me, even though I make a clean and full rebuild of FreeSWITCH. Peter On 09-04-10 14.29, "Michael Jerris" wrote: On Apr 10, 2009, at 4:59 AM, Peter Olsson wrote: > The spidermonkey modules core_db/odbc, curl, socket and teletone > fails to load in Windows. They just return error 1271 (Sym Error). > I'm not sure if this is a known issue, or if just doesn't work in > Windows :) I've been using the latest SVN when trying this. > > Any ideas anyone? :) This was fixed in svn yesterday. 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 !DSPAM:49df3d7c32935147211740! From dujinfang at gmail.com Fri Apr 10 11:38:32 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 11 Apr 2009 02:38:32 +0800 Subject: [Freeswitch-users] skypiax Round Robin interface Message-ID: Hi, I made a patch, so skypiax is possible to do a RR hunt besides the sequential interface ANY. Usage: originate skypiax/RR/other_skype_name sk list http://jira.freeswitch.org/browse/MODENDP-211 From mgg at giagnocavo.net Fri Apr 10 12:47:55 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 10 Apr 2009 15:47:55 -0400 Subject: [Freeswitch-users] How to find/build FreeSWITCH.Managed.dll? In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C67025DE8F196@mse17be1.mse17.exchange.ms> Yes you will need to compile the managed one with C#. It should be enough to go to the freeswitch\src\mod\languages\mod_managed\managed directory and execute msbuild. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Friday, April 10, 2009 4:05 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How to find/build FreeSWITCH.Managed.dll? Hello again! When trying to load the mod_managed module it get an error that it can't find FreeSWITCH.Managed.dll. So my question is simply - where do I find this file, or how do I build it? I'm using the VC++ Express edition when building, so I guess I also have to install the C# edition - will this solve my problem? Sorry for asking stupid questions here - but I've just been playing around with FreeSWITCH for a day or so :) Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 Fri Apr 10 21:37:50 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 10 Apr 2009 23:37:50 -0500 Subject: [Freeswitch-users] announcing mod_cidlookup Message-ID: Another itch scratched. :) I just committed a new module mod_cidlookup. mod_cidlookup allows one to: * lookup number->name mapping in a local database * lookup number->name mapping from a URL * cache the results of the URL lookup in memcache The URL lookup is useful when using third party number to name lookup services. Read more about it at: http://wiki.freeswitch.org/wiki/Mod_cidlookup File a jira or email me if you have issues. -- -Rupa From mszlazak at aol.com Sat Apr 11 00:24:04 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 11 Apr 2009 03:24:04 -0400 Subject: [Freeswitch-users] Kill, close or reset a channel that remains with "show channels count" Message-ID: <8CB888DA0788FAD-8D8-3422@mblk-d38.sysops.aol.com> Initially, "show" "channels count" gives "0 total." I then spawn a session from javascript thus: apiExecute("originate", "{id_name='" + call['Caller Name'] + "',id_number=" + call["Caller Number"] + "}sofia/gateway/spa3102PSTN/" + "1" + call["Caller Number"] + "@10.0.0.5:5061 '&javascript(reminder.js \'${id_name}\' ${id_number})'"); Next, I'd like to automatically spawn another session when my channels show "0 total" with apiExecute("show", "channels count"); I've set up a (loop with a msleep) to check when apiExecute("show", "channels count") becomes zero but it never does after the first call and stays at 1. This seems to "mess-up" making the next call. Thanks. Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/e40e01fe/attachment-0002.html From anthony.minessale at gmail.com Sat Apr 11 09:30:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 11 Apr 2009 11:30:08 -0500 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: References: Message-ID: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> see if it works in latest trunk please. On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem > to detect DTMF. I'm not sure if this is a setting somewhere in the config > files, or if it's a bug. The test scenario is simple - use the default > FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't > detect any DTMF tones. > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/c1f7f9ff/attachment-0002.html From ojab at ojab.ru Sat Apr 11 09:49:43 2009 From: ojab at ojab.ru (ojab) Date: Sat, 11 Apr 2009 20:49:43 +0400 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> References: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> Message-ID: mod_opal in latest trunk is broken http://jira.freeswitch.org/browse/MODOPAL-4 so dtmf definitely doen't work ._. //wbr On Sat, Apr 11, 2009 at 8:30 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > see if it works in latest trunk please. > > > > On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > >> When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem >> to detect DTMF. I'm not sure if this is a setting somewhere in the config >> files, or if it's a bug. The test scenario is simple - use the default >> FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't >> detect any DTMF tones. >> >> Regards, >> >> Peter Olsson >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/cf7f74a9/attachment-0002.html From ojab at ojab.ru Sat Apr 11 09:51:48 2009 From: ojab at ojab.ru (ojab) Date: Sat, 11 Apr 2009 20:51:48 +0400 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: References: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> Message-ID: oops, sorry for the noise, fxd now. On Sat, Apr 11, 2009 at 8:49 PM, ojab wrote: > mod_opal in latest trunk is broken > http://jira.freeswitch.org/browse/MODOPAL-4 so dtmf definitely doen't work > ._. > > //wbr > > > On Sat, Apr 11, 2009 at 8:30 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> see if it works in latest trunk please. >> >> >> >> On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson < >> peter.olsson at visionutveckling.se> wrote: >> >>> When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem >>> to detect DTMF. I'm not sure if this is a setting somewhere in the config >>> files, or if it's a bug. The test scenario is simple - use the default >>> FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't >>> detect any DTMF tones. >>> >>> Regards, >>> >>> Peter Olsson >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> 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/ >> >> 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 >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/f6481835/attachment-0002.html From diego.viola at gmail.com Sat Apr 11 16:50:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 11 Apr 2009 19:50:02 -0400 Subject: [Freeswitch-users] Can't send commands with event socket outbound Message-ID: <86a32abc0904111650n2b72c55eyc33665d014286a77@mail.gmail.com> Hello, I'm testing event socket outbound and whenever I use this: -bash-3.2# nc -v -l 127.0.0.1 8084 I send a call to the socket from my dialplan: I can receive just fine but I can't send events from the nc cli. I tried with: connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n But nothing happens, what's wrong? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/4807ed5b/attachment-0002.html From diego.viola at gmail.com Sat Apr 11 16:59:22 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 11 Apr 2009 19:59:22 -0400 Subject: [Freeswitch-users] Can't send commands with event socket outbound In-Reply-To: <86a32abc0904111650n2b72c55eyc33665d014286a77@mail.gmail.com> References: <86a32abc0904111650n2b72c55eyc33665d014286a77@mail.gmail.com> Message-ID: <86a32abc0904111659m797cb28er4731cbc39f1d0803@mail.gmail.com> Never mind, problem solved. On Sat, Apr 11, 2009 at 7:50 PM, Diego Viola wrote: > Hello, > > I'm testing event socket outbound and whenever I use this: > > -bash-3.2# nc -v -l 127.0.0.1 8084 > > I send a call to the socket from my dialplan: > > > > I can receive just fine but I can't send events from the nc cli. > > I tried with: > > connect\n\n > sendmsg > call-command: execute > execute-app-name: answer\n\n > > But nothing happens, what's wrong? > > Thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/abdb5354/attachment-0002.html From diego.viola at gmail.com Sat Apr 11 19:36:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 11 Apr 2009 22:36:47 -0400 Subject: [Freeswitch-users] Question about play_and_get_digits in mod_dptools Message-ID: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> Hi all, I want to use play_and_get_digits from mod_dptools and have some questions about it. I have used playAndGetDigits() in Lua but I see the syntax in mod_dptools is a bit different and I got a bit confused. I see the syntax in the play_and_get_digits from the mod_dptools is something like this: switch_play_and_get_digits(session, min_digits, max_digits, max_tries, timeout, valid_terminators, prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, sizeof(digit_buffer), digits_regex} Can you please explain to me what the session parameter is? And will this allow me to use a phrase macro so I can call my IVR instead of calling a regular file? This is how I use the playAndGetDigits in Lua: digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", "", "\\d+"); I call a phrase macro instead of playing a file, can I do the same with play_and_get_digits from mod_dptools? and please explain me what the session parameter in play_and_get_digits is. Thanks, Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/4318c6bc/attachment-0002.html From diego.viola at gmail.com Sat Apr 11 19:38:41 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 11 Apr 2009 22:38:41 -0400 Subject: [Freeswitch-users] Question about play_and_get_digits in mod_dptools In-Reply-To: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> References: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> Message-ID: <86a32abc0904111938o2e533481x6fdd0318940ca8a@mail.gmail.com> If you give me some examples of how to use play_and_get_digits in mod_dptools I will document it here. http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_play_and_get_digits Diego On Sat, Apr 11, 2009 at 10:36 PM, Diego Viola wrote: > Hi all, > > I want to use play_and_get_digits from mod_dptools and have some questions > about it. > > I have used playAndGetDigits() in Lua but I see the syntax in mod_dptools > is a bit different and I got a bit confused. > > I see the syntax in the play_and_get_digits from the mod_dptools is > something like this: > > switch_play_and_get_digits(session, min_digits, max_digits, > max_tries, timeout, valid_terminators, > > prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, > sizeof(digit_buffer), digits_regex} > > Can you please explain to me what the session parameter is? And will this > allow me to use a phrase macro so I can call my IVR instead of calling a > regular file? > > This is how I use the playAndGetDigits in Lua: > > digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", > "", "\\d+"); > > I call a phrase macro instead of playing a file, can I do the same with > play_and_get_digits from mod_dptools? and please explain me what the session > parameter in play_and_get_digits is. > > Thanks, > > Diego > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/2607554e/attachment-0002.html From diego.viola at gmail.com Sat Apr 11 20:47:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 11 Apr 2009 23:47:02 -0400 Subject: [Freeswitch-users] Question about play_and_get_digits in mod_dptools In-Reply-To: <86a32abc0904111938o2e533481x6fdd0318940ca8a@mail.gmail.com> References: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> <86a32abc0904111938o2e533481x6fdd0318940ca8a@mail.gmail.com> Message-ID: <86a32abc0904112047k2bc0df91ib6b6a2c703f6acd5@mail.gmail.com> Mike just answered. 23:45 < MikeJ> diegoviola: the syntax is documented 23:45 < MikeJ> SWITCH_ADD_APP(app_interface, "play_and_get_digits", "Play and get Digits", "Play and get Digits", 23:45 < MikeJ> IIII play_and_get_digits_function, " ", SAF_NONE); Thanks guys. Diego On Sat, Apr 11, 2009 at 10:38 PM, Diego Viola wrote: > If you give me some examples of how to use play_and_get_digits in > mod_dptools I will document it here. > > > http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_play_and_get_digits > > Diego > > > On Sat, Apr 11, 2009 at 10:36 PM, Diego Viola wrote: > >> Hi all, >> >> I want to use play_and_get_digits from mod_dptools and have some questions >> about it. >> >> I have used playAndGetDigits() in Lua but I see the syntax in mod_dptools >> is a bit different and I got a bit confused. >> >> I see the syntax in the play_and_get_digits from the mod_dptools is >> something like this: >> >> switch_play_and_get_digits(session, min_digits, max_digits, >> max_tries, timeout, valid_terminators, >> >> prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, >> sizeof(digit_buffer), digits_regex} >> >> Can you please explain to me what the session parameter is? And will this >> allow me to use a phrase macro so I can call my IVR instead of calling a >> regular file? >> >> This is how I use the playAndGetDigits in Lua: >> >> digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", >> "", "\\d+"); >> >> I call a phrase macro instead of playing a file, can I do the same with >> play_and_get_digits from mod_dptools? and please explain me what the session >> parameter in play_and_get_digits is. >> >> Thanks, >> >> Diego >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090411/a1faec3d/attachment-0002.html From pawzlion at gmail.com Sat Apr 11 21:55:14 2009 From: pawzlion at gmail.com (David Robinson) Date: Sun, 12 Apr 2009 14:55:14 +1000 Subject: [Freeswitch-users] Selecting a particular outgoing gateway ? In-Reply-To: References: Message-ID: <6ECDE49F-52F1-48E6-95CA-716A16CF5742@gmail.com> I have configured two providers in sip_profiles/external/example.xml as follows: I have put this in dialplan/defaul.xml to dial out. Regexp matches the first extension if it's a mobile. I want it to route to Pennytelm. voicepulse_pt. Is this how I should be doing this ? I want to specify a different gateway for a different rexep. Please give me some idea what path I should take. David From rupa at rupa.com Sat Apr 11 22:27:27 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Sun, 12 Apr 2009 00:27:27 -0500 Subject: [Freeswitch-users] Selecting a particular outgoing gateway ? In-Reply-To: <6ECDE49F-52F1-48E6-95CA-716A16CF5742@gmail.com> References: <6ECDE49F-52F1-48E6-95CA-716A16CF5742@gmail.com> Message-ID: Sure, you can simply set your regexp in your dialplan. Your example routes numbers starting with 04 to voicepulse_pt 4 or 10 digit numbers go to voicepulse. Now, why you named your gateways voicepulse when neither are using voicepulse, I'll let you ponder. :) If your needs are more complex and/or you have many routes, look into using mod_lcr. I have routes loaded from 2 providers in my tables (teliax and vitelity). phone=> select count(*) from lcr; count ------- 15425 (1 row) Queries against this datbase (postgresql 8.3, using prefix module on a lower end box in the middle of a rsync backup) for lcr info are on the order of 5 milliseconds. http://wiki.freeswitch.org/wiki/Mod_lcr On Sat, Apr 11, 2009 at 11:55 PM, David Robinson wrote: > I have configured two providers in sip_profiles/external/example.xml > as follows: > > > ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? > > ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? > > > I have put this in dialplan/defaul.xml to dial out. Regexp matches the > first extension if it's a mobile. I want it to route to Pennytelm. > voicepulse_pt. > > ? ? > ? ? > ? ? ? data="effective_caller_id_number=11111111111"/> > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? > > ? ? > ? ? > ? ? ? data="effective_caller_id_number=99999999999"/> > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? > > Is this how I should be doing this ? I want to specify a different > gateway for a different rexep. Please give me some idea what path I > should take. > > David > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 From jason at jasonjgw.net Sat Apr 11 22:29:35 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 12 Apr 2009 15:29:35 +1000 Subject: [Freeswitch-users] Selecting a particular outgoing gateway ? In-Reply-To: <6ECDE49F-52F1-48E6-95CA-716A16CF5742@gmail.com> References: <6ECDE49F-52F1-48E6-95CA-716A16CF5742@gmail.com> Message-ID: <20090412052935.GA7458@jdc.jasonjgw.net> David Robinson wrote: > Is this how I should be doing this ? I want to specify a different > gateway for a different rexep. Please give me some idea what path I > should take. Make sure that FreeSWITCH actually reaches your extensions while searching the dial plan. Order is important: if another extension is matched first, and continue="true" is not specified in that extension, your extension will never be invoked. I would also suggest creating a new file in conf/dialplan/default instead of editing the default.xml file, unless of course you want to eliminate some of the extensions provided in default.xml in the sample configurations. Use the sofia status commands from fs_cli to see whether FreeSWITCH is registering to your gateways. for example, sofia status gateway If it doesn't work, the debug logs are in logs/freeswitch.log; read them carefully, as they will usually enable you to pinpoint the problem. To see where your extensions appear in the dial plan, have a look in logs/freeswitch.xml.fsxml, but don't edit that file! I hoe this helps. From diego.viola at gmail.com Sat Apr 11 23:30:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 12 Apr 2009 02:30:47 -0400 Subject: [Freeswitch-users] Question about play_and_get_digits in mod_dptools In-Reply-To: <86a32abc0904112047k2bc0df91ib6b6a2c703f6acd5@mail.gmail.com> References: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> <86a32abc0904111938o2e533481x6fdd0318940ca8a@mail.gmail.com> <86a32abc0904112047k2bc0df91ib6b6a2c703f6acd5@mail.gmail.com> Message-ID: <86a32abc0904112330q7454c12fl99d2c142ea3ec407@mail.gmail.com> I just added this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits Diego On Sat, Apr 11, 2009 at 11:47 PM, Diego Viola wrote: > Mike just answered. > > 23:45 < MikeJ> diegoviola: the syntax is documented > 23:45 < MikeJ> SWITCH_ADD_APP(app_interface, "play_and_get_digits", "Play > and get Digits", "Play and get Digits", > 23:45 < MikeJ> IIII play_and_get_digits_function, " > ", > SAF_NONE); > > Thanks guys. > > Diego > > > On Sat, Apr 11, 2009 at 10:38 PM, Diego Viola wrote: > >> If you give me some examples of how to use play_and_get_digits in >> mod_dptools I will document it here. >> >> >> http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_play_and_get_digits >> >> Diego >> >> >> On Sat, Apr 11, 2009 at 10:36 PM, Diego Viola wrote: >> >>> Hi all, >>> >>> I want to use play_and_get_digits from mod_dptools and have some >>> questions about it. >>> >>> I have used playAndGetDigits() in Lua but I see the syntax in mod_dptools >>> is a bit different and I got a bit confused. >>> >>> I see the syntax in the play_and_get_digits from the mod_dptools is >>> something like this: >>> >>> switch_play_and_get_digits(session, min_digits, max_digits, >>> max_tries, timeout, valid_terminators, >>> >>> prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, >>> sizeof(digit_buffer), digits_regex} >>> >>> Can you please explain to me what the session parameter is? And will this >>> allow me to use a phrase macro so I can call my IVR instead of calling a >>> regular file? >>> >>> This is how I use the playAndGetDigits in Lua: >>> >>> digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", >>> "", "\\d+"); >>> >>> I call a phrase macro instead of playing a file, can I do the same with >>> play_and_get_digits from mod_dptools? and please explain me what the session >>> parameter in play_and_get_digits is. >>> >>> Thanks, >>> >>> Diego >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090412/86a64435/attachment-0002.html From diego.viola at gmail.com Sun Apr 12 00:50:32 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 12 Apr 2009 03:50:32 -0400 Subject: [Freeswitch-users] Question about play_and_get_digits in mod_dptools In-Reply-To: <86a32abc0904112330q7454c12fl99d2c142ea3ec407@mail.gmail.com> References: <86a32abc0904111936s6be0330ch50ce05ea08dcce35@mail.gmail.com> <86a32abc0904111938o2e533481x6fdd0318940ca8a@mail.gmail.com> <86a32abc0904112047k2bc0df91ib6b6a2c703f6acd5@mail.gmail.com> <86a32abc0904112330q7454c12fl99d2c142ea3ec407@mail.gmail.com> Message-ID: <86a32abc0904120050x4c6ddfe5hbf10f26be079b2d1@mail.gmail.com> I wish I would have seen this before =D 03:36 < Math> diegoviola: show application [appname] will show you any syntax btw Diego On Sun, Apr 12, 2009 at 2:30 AM, Diego Viola wrote: > I just added this: > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits > > Diego > > > On Sat, Apr 11, 2009 at 11:47 PM, Diego Viola wrote: > >> Mike just answered. >> >> 23:45 < MikeJ> diegoviola: the syntax is documented >> 23:45 < MikeJ> SWITCH_ADD_APP(app_interface, "play_and_get_digits", "Play >> and get Digits", "Play and get Digits", >> 23:45 < MikeJ> IIII play_and_get_digits_function, " >> ", >> SAF_NONE); >> >> Thanks guys. >> >> Diego >> >> >> On Sat, Apr 11, 2009 at 10:38 PM, Diego Viola wrote: >> >>> If you give me some examples of how to use play_and_get_digits in >>> mod_dptools I will document it here. >>> >>> >>> http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_play_and_get_digits >>> >>> Diego >>> >>> >>> On Sat, Apr 11, 2009 at 10:36 PM, Diego Viola wrote: >>> >>>> Hi all, >>>> >>>> I want to use play_and_get_digits from mod_dptools and have some >>>> questions about it. >>>> >>>> I have used playAndGetDigits() in Lua but I see the syntax in >>>> mod_dptools is a bit different and I got a bit confused. >>>> >>>> I see the syntax in the play_and_get_digits from the mod_dptools is >>>> something like this: >>>> >>>> switch_play_and_get_digits(session, min_digits, max_digits, >>>> max_tries, timeout, valid_terminators, >>>> >>>> prompt_audio_file, bad_input_audio_file, var_name, digit_buffer, >>>> sizeof(digit_buffer), digits_regex} >>>> >>>> Can you please explain to me what the session parameter is? And will >>>> this allow me to use a phrase macro so I can call my IVR instead of calling >>>> a regular file? >>>> >>>> This is how I use the playAndGetDigits in Lua: >>>> >>>> digits = session:playAndGetDigits(2, 5, 3, 7000, "#", "phrase:rac_demo", >>>> "", "\\d+"); >>>> >>>> I call a phrase macro instead of playing a file, can I do the same with >>>> play_and_get_digits from mod_dptools? and please explain me what the session >>>> parameter in play_and_get_digits is. >>>> >>>> Thanks, >>>> >>>> Diego >>>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090412/2bb937dc/attachment-0002.html From peter.olsson at visionutveckling.se Sun Apr 12 04:28:50 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 12 Apr 2009 13:28:50 +0200 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> Message-ID: Thanks, I tried latest trunk, but still no success.. :( Peter On 09-04-11 18.30, "Anthony Minessale" wrote: see if it works in latest trunk please. On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson wrote: When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem to detect DTMF. I'm not sure if this is a setting somewhere in the config files, or if it's a bug. The test scenario is simple - use the default FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't detect any DTMF tones. Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 e.schmidbauer at gmail.com Sun Apr 12 10:11:59 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Sun, 12 Apr 2009 13:11:59 -0400 Subject: [Freeswitch-users] ekiga and freeswitch Message-ID: <2cef777b0904121011q1e3df133xf2814d609effbbd7@mail.gmail.com> I am running FreeSWITCH Version 1.0.3 on a centos 5.2 x64 server and ekiga 3.2.0 on a client computer. I am able to register the ekiga client with the freeswitch server but when I dial an extension to join a conference I get the following error message: [ERR] switch_core_io.c:327 switch_core_session_read_frame() Codec RAW Signed Linear (16 bit) decoder error! I have the celt codec loaded on both the server and the client computer. This is the full output when dialing an extension: 2009-04-12 12:56:34 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing joseph schmidbauer->4800 in context default 2009-04-12 12:56:34 [INFO] mod_sofia.c:1301 sofia_receive_message() Asked to send early media by sofia/internal/joe at 192.168.1.125 2009-04-12 12:56:34 [INFO] mod_sofia.c:1342 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1239529036 1239529037 IN IP4 192.168.1.125 s=FreeSWITCH c=IN IP4 192.168.1.125 t=0 0 m=audio 26358 RTP/AVP 116 101 a=rtpmap:116 CELT/48000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv 2009-04-12 12:56:34 [NOTICE] mod_sofia.c:1345 sofia_receive_message() Pre-Answer sofia/internal/joe at 192.168.1.125! 2009-04-12 12:56:34 [NOTICE] mod_conference.c:1934 conference_loop_output() Channel [sofia/internal/joe at 192.168.1.125] has been answered warning: decode error [ERR] switch_core_io.c:327 switch_core_session_read_frame() Codec RAW Signed Linear (16 bit) decoder error! [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/internal/joe at 192.168.1.125 [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-12 13:05:14 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 4 (sofia/internal/joe at 192.168.1.125) Ended 2009-04-12 13:05:14 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/joe at 192.168.1.125 [CS_HANGUP] Not sure if this is a bug in the program or just in my setup. I've tried using the svn version of freeswitch (as of yesterday) and i got the exact same error. Any input would be appreciated. Thanks! From brian at freeswitch.org Sun Apr 12 10:34:29 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 12 Apr 2009 12:34:29 -0500 Subject: [Freeswitch-users] ekiga and freeswitch In-Reply-To: <2cef777b0904121011q1e3df133xf2814d609effbbd7@mail.gmail.com> References: <2cef777b0904121011q1e3df133xf2814d609effbbd7@mail.gmail.com> Message-ID: <159790A6-1FD5-4831-858B-47666E45C9B3@freeswitch.org> Collect a full sip trace and FULL console debug. Put it on our pastebin... Chances are Ekiga is doing something stupid... it usually does silly things. Also are you on SVN trunk? /b On Apr 12, 2009, at 12:11 PM, e schmidbauer wrote: > Not sure if this is a bug in the program or just in my setup. > I've tried using the svn version of freeswitch (as of yesterday) and i > got the exact same error. Any input would be appreciated. Thanks! From eric at rf.com Sun Apr 12 11:05:15 2009 From: eric at rf.com (Eric Chamberlain) Date: Sun, 12 Apr 2009 11:05:15 -0700 Subject: [Freeswitch-users] Can gateways be configured on a per user basis? Message-ID: <0FDF3663-4CC4-424B-99DE-143039A2213D@rf.com> Hello, I'm exploring the capabilities of FreeSwitch and have some questions: Can gateways be configured on a per user basis? A user needs to be limited to only their own gateways and other users can't user their gateway. Something like: User 1 has two gateways - VoIP provider A and VoIP provider B User 2 also has two gateways - VoIP provider A and VoIP provider C User 3 has ten gateways User 4 has one gateway etc. Is it possible to have some gateway settings global and some user specific, say proxy and port info global, but username and password are specific to each user? Is it also possible to specify on a per user basis whether the user's gateway will register for the provider? Using the example above, User 1 and User 2 would both register with VoIP provider A. And can that registration happen even if User 1 or User 2 don't have any SIP endpoints registered with FreeSwitch? -- Eric Chamberlain From nicolas at medularis.com Sun Apr 12 22:00:24 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Mon, 13 Apr 2009 01:00:24 -0400 Subject: [Freeswitch-users] Replace sqlite with couchDB? Message-ID: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> Hi, I am not very familiar with FS internals, but I recently found this "new" db engine called couchDB. Looks pretty interesting, and its main focus is scalability. Has anybody played with couchDB? does it make sense to replace sqlite with couchDB in FS? Here's a link to the project homepage: - http://couchdb.apache.org/ And here's a video of a presentation given by one of the lead programmers: - http://www.vimeo.com/1992869 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/6e89afdb/attachment-0002.html From mattdfong at gmail.com Sun Apr 12 22:06:46 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Mon, 13 Apr 2009 12:06:46 +0700 Subject: [Freeswitch-users] Replace sqlite with couchDB? In-Reply-To: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> References: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> Message-ID: <4256bf830904122206v156a826dm43084ba30a315bd0@mail.gmail.com> Hi Nicolas, Just off the top of my head, but I think couchDB is rather large compared to sqlite, and I think it's also geared more towards storing dynamic datasets...rather ones that can be structured...like FS calling data can. But I might be wrong :) your buddy. --matt On Mon, Apr 13, 2009 at 12:00 PM, Nicolas Brenner wrote: > Hi, I am not very familiar with FS internals, but I recently found this > "new" db engine called couchDB. Looks pretty interesting, and its main focus > is scalability. > Has anybody played with couchDB? does it make sense to replace sqlite with > couchDB in FS? > > Here's a link to the project homepage: > - http://couchdb.apache.org/ > > And here's a video of a presentation given by one of the lead programmers: > - http://www.vimeo.com/1992869 > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/d382f8a5/attachment-0002.html From nicolas at medularis.com Sun Apr 12 22:33:26 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Mon, 13 Apr 2009 01:33:26 -0400 Subject: [Freeswitch-users] Replace sqlite with couchDB? In-Reply-To: <4256bf830904122206v156a826dm43084ba30a315bd0@mail.gmail.com> References: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> <4256bf830904122206v156a826dm43084ba30a315bd0@mail.gmail.com> Message-ID: <1b46b4e80904122233j21201b2ela84c72fd1c597dcc@mail.gmail.com> Well, if it's too large compared to sqlite maybe it doesn't make sense. But I was thinking calling data is not always fixed. Depending on what you use FS for, you might want to get a CDR with many different data linked to each call, even different kinds of data linked to different calls, that would make each call very different and variable in its structure, which would fit a document db model. Thinking a bit more now, since couchdb is a document-based DB, it might be good for configuration-generating applications, like the ones consumed by xml_curl. These are external applications, yet they are still very closely related to FS, and might be able to benefit from using something like couchdb. On Mon, Apr 13, 2009 at 1:06 AM, Matthew Fong wrote: > Hi Nicolas, > Just off the top of my head, but I think couchDB is rather large compared > to sqlite, and I think it's also geared more towards > storing dynamic datasets...rather ones that can be structured...like FS > calling data can. > > But I might be wrong :) > your buddy. > > --matt > > On Mon, Apr 13, 2009 at 12:00 PM, Nicolas Brenner wrote: > >> Hi, I am not very familiar with FS internals, but I recently found this >> "new" db engine called couchDB. Looks pretty interesting, and its main focus >> is scalability. >> Has anybody played with couchDB? does it make sense to replace sqlite with >> couchDB in FS? >> >> Here's a link to the project homepage: >> - http://couchdb.apache.org/ >> >> And here's a video of a presentation given by one of the lead programmers: >> - http://www.vimeo.com/1992869 >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/8f87b080/attachment-0002.html From jason at jasonjgw.net Sun Apr 12 22:52:36 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 13 Apr 2009 15:52:36 +1000 Subject: [Freeswitch-users] Replace sqlite with couchDB? In-Reply-To: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> References: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> Message-ID: <20090413055236.GA19344@jdc.jasonjgw.net> Nicolas Brenner wrote: > Hi, I am not very familiar with FS internals, but I recently found this > "new" db engine called couchDB. Looks pretty interesting, and its main focus > is scalability. > Has anybody played with couchDB? does it make sense to replace sqlite with > couchDB in FS? I think a lot of people would object to replacing a small database such as SQLite, which is easily integrated into the FreeSWITCh source code, with an Erlang application. Somehow, I don't see FreeSWITCH users accepting all of the dependencies that would bring, unless they're already using the Erlang module for other reasons. However, if it would be of benefit to Erlang users, I'm sure the FreeSWITCH developers would gladly accept a module. There are lots of databases out there, for example, http://monetdb.cwi.nl/ to mention just one that a Web search located for me. Which ones get supported depends on whether anyone is sufficiently interested to write modules for them. PostGRESQL and MySQL are already on the list, notably. From zhaoxxqq at 163.com Mon Apr 13 03:26:47 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Mon, 13 Apr 2009 18:26:47 +0800 Subject: [Freeswitch-users] Noise for dial out conference Message-ID: <200904131826464590579@163.com> Hello, I use below to realize call out conference: conference testconf bgdial {originate_timeout=30}sofia/default/1001 at 192.168.0.72 1234567890 FreeSWITCH_Conference" conference testconf bgdial {originate_timeout=30}sofia/default/1002 at 192.168.0.170 1234567890 FreeSWITCH_Conference" but when the second phone is connected. there are big noise in the phone, Can anyone help me to solve it? zhao xiaoqiang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/3c07e276/attachment-0002.html From uv at yuvalhertzog.com Mon Apr 13 04:32:12 2009 From: uv at yuvalhertzog.com (UV) Date: Mon, 13 Apr 2009 21:32:12 +1000 Subject: [Freeswitch-users] Skypiax as a windows service Message-ID: <0D1E9E22CCAC4F98ADA9863FDFF7FB85@UVix> Great work on Skypiax, Giovanni. We've tested it in our lab for sometime and it works very well. Unfortunately, when we tried deploying it on a production environment (running Win2K3 server farm), we ran into a barrier: 1. FS is running as terminal server console application (to be easily maintained remotely by RDP) 2. This is because Win2K does not allow RDP to access system console (session /userid 0) 3. Skype does not work on terminal server due to a well known disappearing audio drivers problem, therefore it has to run either as a console or a service (both on session 0). 4. FS can run well as a windows service 5. Skypiax seem to load as service, but it can't find the skype client and exit with the following error: 2009-04-13 20:54:14 [ERR] mod_skypiax.c:990 load_config() rev 13006M[00000000|37 ][ERRORA 990 ][skype_user ][-1, 0, 0] Failed to connect to a SKYPE API for interface_id=1, no SKYPE client running, please (re)start Skype client. Skypiax exiting This situation prevents me to run skypiax in production. I understand from the wiki page that windows service is not done yet - so I presume this is a predicted outcome. Any idea when and if this is planned to be implemented? Keep up the good work! Cheers, UV -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/a20a7842/attachment-0002.html From yudha2008 at gmail.com Mon Apr 13 06:09:38 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 13 Apr 2009 18:39:38 +0530 Subject: [Freeswitch-users] Mod_java loading error Message-ID: Hi, I have loaded the java module in freeswitch. But when i run freeswitch in console i get this error. 2009-04-13 17:38:33 [NOTICE] modjava.c:244 mod_java_load() Java Framework Loading... 2009-04-13 17:38:33 [ERR] modjava.c:133 load_config() Error loading /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so 2009-04-13 17:38:33 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** Can any one specify what is the error. I am using Freeswitch 1.0.2 in CentOS 5.2. Thanks in advance. -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/608abf4b/attachment-0002.html From brian at freeswitch.org Mon Apr 13 06:15:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 13 Apr 2009 08:15:39 -0500 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: Message-ID: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> You should update to SVN trunk and try again. /b On Apr 13, 2009, at 8:09 AM, Baskar wrote: > Hi, > > I have loaded the java module in freeswitch. But when i run > freeswitch in console i get this error. > > 2009-04-13 17:38:33 [NOTICE] modjava.c:244 mod_java_load() Java > Framework Loading... > 2009-04-13 17:38:33 [ERR] modjava.c:133 load_config() Error loading / > usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so > 2009-04-13 17:38:33 [CRIT] switch_loadable_module.c:839 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_java.so > **Module load routine returned an error** > > Can any one specify what is the error. I am using Freeswitch 1.0.2 > in CentOS 5.2. > > Thanks in advance. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/3bacc781/attachment-0002.html From grevenx at me.com Mon Apr 13 08:13:39 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Mon, 13 Apr 2009 17:13:39 +0200 Subject: [Freeswitch-users] Replace sqlite with couchDB? In-Reply-To: <20090413055236.GA19344@jdc.jasonjgw.net> References: <1b46b4e80904122200n6defcb69r1ce5ca5753cb5b39@mail.gmail.com> <20090413055236.GA19344@jdc.jasonjgw.net> Message-ID: <92AD0C5B-366C-4947-B6ED-9F10C6EBADB8@me.com> The only part I see fit for integration with CouchDB is for storing CDR documents. This kind of database is imho best-used for storing large sets of data, in a document structure. I don't think the FS config fits this description, since the amount of config documents are typically not "large". You can also look at related distributed systems like Hadoop/Hbase, which could be good to store CDRs in. It's been a couple of months since I researched these systems, but I think it's possible to enable an HTTP REST interface for both, so you could use the built-in feature for posting CDRs to a HTTP server. Best regards, Even Andr? On 13. april. 2009, at 07.52, Jason White wrote: > Nicolas Brenner wrote: >> Hi, I am not very familiar with FS internals, but I recently found >> this >> "new" db engine called couchDB. Looks pretty interesting, and its >> main focus >> is scalability. >> Has anybody played with couchDB? does it make sense to replace >> sqlite with >> couchDB in FS? > > I think a lot of people would object to replacing a small database > such as > SQLite, which is easily integrated into the FreeSWITCh source code, > with an > Erlang application. Somehow, I don't see FreeSWITCH users accepting > all of the > dependencies that would bring, unless they're already using the > Erlang module > for other reasons. However, if it would be of benefit to Erlang > users, I'm > sure the FreeSWITCH developers would gladly accept a module. > > There are lots of databases out there, for example, http://monetdb.cwi.nl/ > to > mention just one that a Web search located for me. > > Which ones get supported depends on whether anyone is sufficiently > interested > to write modules for them. > > PostGRESQL and MySQL are already on the list, notably. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Mon Apr 13 08:25:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Apr 2009 10:25:22 -0500 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: References: <191c3a030904110930q4ba4978bged3a1cd26e0c665e@mail.gmail.com> Message-ID: <191c3a030904130825q32f59ddagf19700aea64e5dfe@mail.gmail.com> which rev was "latest" for you? I have confirmation that it is indeed working. On Sun, Apr 12, 2009 at 6:28 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Thanks, > > I tried latest trunk, but still no success.. :( > > Peter > > > On 09-04-11 18.30, "Anthony Minessale" > wrote: > > see if it works in latest trunk please. > > > On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem > to detect DTMF. I'm not sure if this is a setting somewhere in the config > files, or if it's a bug. The test scenario is simple - use the default > FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't > detect any DTMF tones. > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/11bd27c9/attachment-0002.html From peter.olsson at visionutveckling.se Mon Apr 13 08:29:59 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 13 Apr 2009 17:29:59 +0200 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... Message-ID: I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson From anthony.minessale at gmail.com Mon Apr 13 08:33:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Apr 2009 10:33:08 -0500 Subject: [Freeswitch-users] Can gateways be configured on a per user basis? In-Reply-To: <0FDF3663-4CC4-424B-99DE-143039A2213D@rf.com> References: <0FDF3663-4CC4-424B-99DE-143039A2213D@rf.com> Message-ID: <191c3a030904130833hf3c73a5meca407f98f77b6d0@mail.gmail.com> You can store gateway xml in both a user and in the sofia.conf, we don't enforce which ones who can use because that would be a function of the dialplan but you can certainly store them in your configs that way and set the names of a user's specifc gateways in a variable that would be present on each inbound call. On Sun, Apr 12, 2009 at 1:05 PM, Eric Chamberlain wrote: > Hello, > > I'm exploring the capabilities of FreeSwitch and have some questions: > > Can gateways be configured on a per user basis? > > A user needs to be limited to only their own gateways and other users > can't user their gateway. > > Something like: > > User 1 has two gateways - VoIP provider A and VoIP provider B > User 2 also has two gateways - VoIP provider A and VoIP provider C > User 3 has ten gateways > User 4 has one gateway > etc. > > Is it possible to have some gateway settings global and some user > specific, say proxy and port info global, but username and password > are specific to each user? > > Is it also possible to specify on a per user basis whether the user's > gateway will register for the provider? Using the example above, User > 1 and User 2 would both register with VoIP provider A. > And can that registration happen even if User 1 or User 2 don't have > any SIP endpoints registered with FreeSwitch? > > -- > Eric Chamberlain > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/f6a574e9/attachment-0002.html From peter.olsson at visionutveckling.se Mon Apr 13 08:42:56 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 13 Apr 2009 17:42:56 +0200 Subject: [Freeswitch-users] mod_opal and DTMF... In-Reply-To: <191c3a030904130825q32f59ddagf19700aea64e5dfe@mail.gmail.com> Message-ID: Anthony, Please read my comments on Jira cases MODOPAL-3 and MODOPAL-5. In MODOPAL-3 I've also attached a patch to get DTMF working better (handle both Tone and String input), and also increasing the stability of mod_opal. But the problem also is connected to MODOPAL-5, which I created just now. It seems to be a codec issue when using A-Law, which I didn't found out until now - that's why the in-band DTMF detection didn't work. But I have DTMF working right now, but it's a bit improved with my patch :) Regards, Peter On 09-04-13 17.25, "Anthony Minessale" wrote: which rev was "latest" for you? I have confirmation that it is indeed working. On Sun, Apr 12, 2009 at 6:28 AM, Peter Olsson wrote: Thanks, I tried latest trunk, but still no success.. :( Peter On 09-04-11 18.30, "Anthony Minessale" wrote: see if it works in latest trunk please. On Fri, Apr 10, 2009 at 3:46 AM, Peter Olsson wrote: When dialing in to FreeSWITCH from a h323 client, FreeSWITCH doesn't seem to detect DTMF. I'm not sure if this is a setting somewhere in the config files, or if it's a bug. The test scenario is simple - use the default FreeSWITCH config, dial in to voicemail (4000) and try to log in. It doesn't detect any DTMF tones. Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 Mon Apr 13 08:54:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Apr 2009 10:54:56 -0500 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: References: Message-ID: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> 1) When you say latest, which rev does that mean? we change revs pretty often. 2) Do you have a minimal script that reproduces your issue. 3) is there a reason you cannot just session.execute("bridge", dest); instead of doing it manually (which is a process not for the faint at heart)? On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > I have two problems that I haven't been able to solve. I've done the same > tests in both javascript, and in .NET. > > The two scripts are pretty simple, they just answer an incomming call, > creates a new session, wait for an answer on the second call leg, and then > bridge the two channels together. > > In both cases everything works just fine, but the audio is distorted. The > destination I'm calling is "loopback/5000" - the sample IVR application > included in FreeSWITCH. I first thought it was a codec issue, but even after > trying to switch to different codecs the problem was the same. It more > sounds like it's a timestamping issue - the voice is not distorted enough to > be a bad codec, but it reads way to fast (mayby twice the "normal" speed). > When doing a direct transfer() to the other destination this works just > fine, but I need to be able to have some extra logic to tell if the > destination is available or not. > > The second problem occurs only in .NET. After doing this sample there is as > loopback channel still hanging around. It seems like the call creates a > loopback-a and loopback-b, the loopback-b dissapears as it should (when the > call has been disconnected), but the other one stays there. When doing the > same in javascript this doesn't seem to occur. > > I'm using the latest SVN trunk, and my OS is Windows XP. > > I found bug FSCORE-349 in Jira, which seems to point in to the direction > that there might be a bug with the loopback channels in some cases, but I > could not find anything about the audio which plays too fast. > > Has anyone else experienced this? > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/21fe407e/attachment-0002.html From peter.olsson at visionutveckling.se Mon Apr 13 09:21:22 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 13 Apr 2009 18:21:22 +0200 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> Message-ID: 1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available.. 2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg. 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. :) What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected? --- if (session.ready()) { session.answer(); new_session = new Session("loopback/5000", session); new_session.waitForAnswer(); bridge(session, new_session); // Not sure if this is needed - I've tried with it both enabled and disabled session.hangup(); new_session.hangup(); } Peter On 09-04-13 17.54, "Anthony Minessale" wrote: 1) When you say latest, which rev does that mean? we change revs pretty often. 2) Do you have a minimal script that reproduces your issue. 3) is there a reason you cannot just session.execute("bridge", dest); instead of doing it manually (which is a process not for the faint at heart)? On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson wrote: I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 berni at birkenwald.de Mon Apr 13 09:35:06 2009 From: berni at birkenwald.de (Bernhard Schmidt) Date: Mon, 13 Apr 2009 16:35:06 +0000 (UTC) Subject: [Freeswitch-users] Can't call registered user in internal-ipv6 profile Message-ID: Hi, probably a pretty easy problem, but I can't figure it out nevertheless. I'm still experimenting with FreeSwitch (SVN trunk, about two weeks old), mainly due to the IPv6 SIP support. I'm pretty much running the default configuration included in the sourcetree. Now I've hit the following problem: I have two phones registered, one Snom with extension 1000 on IPv4 (profile internal), one SIP Communicator with extension 1002 on IPv6 (profile internal-ipv6). I can call the Snom just fine (from the SIP Communicator as well as from outside or the CLI), but not the SIP Communicator. EXECUTE sofia/internal/1000 at obelix.oms16.birkenwald.de bridge(user/1002 at 172.16.1.69) 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:1077 switch_ivr_originate() variable string 0 = [presence_id=1002 at 172.16.1.69] 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 switch_ivr_originate() Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 switch_ivr_originate() Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 switch_ivr_originate() Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2009-04-13 18:23:45 [INFO] mod_dptools.c:2051 audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED It works when I explicitly specify the internal-ipv6 profile for the outgoing call like this: but that isn't really what I want. What are the quirks I need to add to have "user/@" search both profiles? I already set force-register-domain on the profile, but I don't think that is what I'm looking for. Bernhard From fialkam at gmail.com Mon Apr 13 09:56:11 2009 From: fialkam at gmail.com (Martin Fiala) Date: Mon, 13 Apr 2009 18:56:11 +0200 Subject: [Freeswitch-users] SIP switching made simple? Message-ID: Hello. I'm trying to use freeswitch, was able to compile it without problems, which is very nice. Then studying the configurations etc., I managed to set up SIP accounts those register properly. But now, if I want to call one registered account from the other one, I get error 404 - not found. I tried to set up a minimalistic dialplan using xml syntax as well as asterisk syntax but neither worked for me. I changed just a few thing, I'll list them later. I'm trying to make calls using ip addresses and ports instead of domain names.. This is the error freeswitch outputs: 2009-04-13 18:35:48 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/02 at 192.168.2.100 [19bad83a-ec9a-4b59-8457-cd76f1eaef65] 2009-04-13 18:35:48 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 02->01 in context default 2009-04-13 18:35:48 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/02 at 192.168.2.100 to enum[01 at default] 2009-04-13 18:35:50 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting 2009-04-13 18:35:50 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup sofia/internal/02 at 192.168.2.100 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 1 (sofia/internal/02 at 192.168.2.100) Ended 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/02 at 192.168.2.100 [CS_HANGUP] My users are added in file users.xml in directory/ : I've added the file dialplan/default/000_default.xml with contents: That's from sample configs, I wonder, if the IP address can be used like that. I understand it that way, the ip address specified is of registrar server. I've added the port as I'm testing it on local loop and thus am running different sip services on the same ip (freeswitch and calling softfones). Is that ok? extensions.conf I've tried to use: [default] ; Things you're used to.... ;exten => music,n,Dial(SIP/1234 at conference.freeswitch.org|120) ;exten => _1XXXXX,n,set(cool=${EXTEN}) ;exten => _1XXXXX,n,set(myvar=true) ;exten => _1XXXXX,n,Goto(default|music) ;exten => 2137991400/1000,n,Goto(default|music) ; Some new magic you can do.... ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,enum($1) ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,bridge(${enum_auto_route}) ; instead of exten, put anything about the call you would rather match on. ; either the names of a field in caller_profile or a string of variables to expand. ;caller_id_number => 2137991400,n,Goto(default|music) ;${sip_from_user} => bill,n,Goto(default|music) [pbx] exten => 01,1,Dial(SIP/01,20) exten => 02,1,Dial(SIP/02,20) When using extensions.conf I've changed this line in sip_profiles/internal.xml from: to I didn't make any other changes in that file. I didn't change anything else. I'm trying to use two sip phones - one using port 6001 (user "01") and the other one 5000 (user "02"). After registration succeeds, calling this sip uri : sip:01 at 192.168.2.100:5060, where 192.168.2.100:5060 is IP:PORT of freeswitch (the IP is same for softphones.. the same machine). Thanks for any help. Fiala From msc at freeswitch.org Mon Apr 13 10:13:27 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Apr 2009 10:13:27 -0700 Subject: [Freeswitch-users] Noise for dial out conference In-Reply-To: <200904131826464590579@163.com> References: <200904131826464590579@163.com> Message-ID: <87f2f3b90904131013y1de87de8v5ab9679d27f68df7@mail.gmail.com> Are both of these phones on the same LAN as FreeSWITCH? What kind of phones are they? Also, can you reverse the dialing order and reproduce the symptoms? Just curious if it's always the same phone causing the noise or if it is always the second phone connected, regardless of which phone. -MC On Mon, Apr 13, 2009 at 3:26 AM, zhaoxxqq wrote: > Hello, > > I use below to realize call out conference: > conference testconf bgdial {originate_timeout=30}sofia/default/ > 1001 at 192.168.0.72 1234567890 FreeSWITCH_Conference" > conference testconf bgdial {originate_timeout=30}sofia/default/ > 1002 at 192.168.0.170 1234567890 FreeSWITCH_Conference" > but when the second phone is connected. there are big noise in the phone, > Can anyone help me to solve it? > > zhao xiaoqiang > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/91ff7ff2/attachment-0002.html From brian at freeswitch.org Mon Apr 13 10:17:29 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 13 Apr 2009 12:17:29 -0500 Subject: [Freeswitch-users] Can't call registered user in internal-ipv6 profile In-Reply-To: References: Message-ID: <04D731AC-5A70-4515-8ABA-B432D18871FA@freeswitch.org> user/ uses the dial-string in the domain to find the user on the internal profile by default.. so to call someone registered via ipv6 you'll need to put a dial-string param in the user to find them on the internal-ipv6 profile. /b On Apr 13, 2009, at 11:35 AM, Bernhard Schmidt wrote: > Hi, > > probably a pretty easy problem, but I can't figure it out > nevertheless. > > I'm still experimenting with FreeSwitch (SVN trunk, about two weeks > old), mainly due to the IPv6 SIP support. I'm pretty much running the > default configuration included in the sourcetree. Now I've hit the > following problem: > > I have two phones registered, one Snom with extension 1000 on IPv4 > (profile internal), one SIP Communicator with extension 1002 on IPv6 > (profile internal-ipv6). I can call the Snom just fine (from the SIP > Communicator as well as from outside or the CLI), but not the SIP > Communicator. > > EXECUTE sofia/internal/1000 at obelix.oms16.birkenwald.de bridge(user/1002 at 172.16.1.69 > ) > 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:1077 > switch_ivr_originate() variable string 0 = [presence_id=1002 at 172.16.1.69 > ] > 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 > switch_ivr_originate() Cannot create outgoing channel of type > [error] cause: [USER_NOT_REGISTERED] > 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 > switch_ivr_originate() Originate Resulted in Error Cause: 606 > [USER_NOT_REGISTERED] > 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 > switch_ivr_originate() Cannot create outgoing channel of type [user] > cause: [USER_NOT_REGISTERED] > 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 > switch_ivr_originate() Originate Resulted in Error Cause: 606 > [USER_NOT_REGISTERED] > 2009-04-13 18:23:45 [INFO] mod_dptools.c:2051 > audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED > > It works when I explicitly specify the internal-ipv6 profile for the > outgoing call like this: > > > > but that isn't really what I want. > > What are the quirks I need to add to have "user/@" > search both profiles? I already set force-register-domain on the > profile, but I don't think that is what I'm looking for. > > Bernhard > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/0a14f6f6/attachment-0002.html From msc at freeswitch.org Mon Apr 13 10:25:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Apr 2009 10:25:16 -0700 Subject: [Freeswitch-users] SIP switching made simple? In-Reply-To: References: Message-ID: <87f2f3b90904131025k635be5faqa46cc9228381aedd@mail.gmail.com> I highly recommend that you set aside this endeavor for the time being and use the default configuration. Once you get familiar with the default config then you'll realize how to make changes to registered users and to the dialplan. Don't let the size of the default configuration scare you off. It is very well designed, and much of it is compartmentalized, which means you can changes in a single file without affecting the rest of the configuration. Now for the usual questions: What platform are you on? Linux? Did you use SVN? (We highly recommend using SVN) Have you seen the wiki pages on installing FS? If you're running on Linux then I recommend a clean install using the method documented here: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install Let us know how it goes. -MC On Mon, Apr 13, 2009 at 9:56 AM, Martin Fiala wrote: > Hello. > > I'm trying to use freeswitch, was able to compile it without problems, > which is very nice. Then studying the configurations etc., I managed > to set up SIP accounts those register properly. But now, if I want to > call one registered account from the other one, I get error 404 - not > found. I tried to set up a minimalistic dialplan using xml syntax as > well as asterisk syntax but neither worked for me. I changed just a > few thing, I'll list them later. I'm trying to make calls using ip > addresses and ports instead of domain names.. > > This is the error freeswitch outputs: > 2009-04-13 18:35:48 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel sofia/internal/02 at 192.168.2.100 > [19bad83a-ec9a-4b59-8457-cd76f1eaef65] > 2009-04-13 18:35:48 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 02->01 in context default > 2009-04-13 18:35:48 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() Transfer sofia/internal/02 at 192.168.2.100 > to enum[01 at default] > 2009-04-13 18:35:50 [INFO] switch_core_state_machine.c:122 > switch_core_standard_on_routing() No Route, Aborting > 2009-04-13 18:35:50 [NOTICE] switch_core_state_machine.c:123 > switch_core_standard_on_routing() Hangup > sofia/internal/02 at 192.168.2.100 [CS_ROUTING] [NO_ROUTE_DESTINATION] > 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 1 > (sofia/internal/02 at 192.168.2.100) Ended > 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel > sofia/internal/02 at 192.168.2.100 [CS_HANGUP] > > My users are added in file users.xml in directory/ : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I've added the file dialplan/default/000_default.xml with contents: > > > > /> > > > That's from sample configs, I wonder, if the IP address can be used > like that. I understand it that way, the ip address specified is of > registrar server. I've added the port as I'm testing it on local loop > and thus am running different sip services on the same ip (freeswitch > and calling softfones). Is that ok? > > > > extensions.conf I've tried to use: > [default] > > ; Things you're used to.... > ;exten => music,n,Dial(SIP/1234 at conference.freeswitch.org|120) > > ;exten => _1XXXXX,n,set(cool=${EXTEN}) > ;exten => _1XXXXX,n,set(myvar=true) > ;exten => _1XXXXX,n,Goto(default|music) > ;exten => 2137991400/1000,n,Goto(default|music) > > > ; Some new magic you can do.... > ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,enum($1) > ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,bridge(${enum_auto_route}) > > ; instead of exten, put anything about the call you would rather match on. > ; either the names of a field in caller_profile or a string of > variables to expand. > ;caller_id_number => 2137991400,n,Goto(default|music) > ;${sip_from_user} => bill,n,Goto(default|music) > > [pbx] > exten => 01,1,Dial(SIP/01,20) > exten => 02,1,Dial(SIP/02,20) > > > > > When using extensions.conf I've changed this line in > sip_profiles/internal.xml from: > > to > > I didn't make any other changes in that file. > > > I didn't change anything else. > > I'm trying to use two sip phones - one using port 6001 (user "01") > and the other one 5000 (user "02"). After registration succeeds, > calling this sip uri : sip:01 at 192.168.2.100:5060, where > 192.168.2.100:5060 is IP:PORT of freeswitch (the IP is same for > softphones.. the same machine). > > Thanks for any help. > Fiala > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/b659f281/attachment-0002.html From anthony.minessale at gmail.com Mon Apr 13 11:37:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Apr 2009 13:37:43 -0500 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> Message-ID: <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> see how it works in latest trunk 13011 nontheless you can just say session.execute("bridge", "loopback/5000"); and get the same result without touching that other channel. when the call fails, you will have an originate_disposition variable in session you can check. On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > 1. The latest trunk I've tried with is 13008. Since I'm not doing > anything for production yet (just testing/evaluating), so I tend to update > as soon as there is new version available.. > 2. Yep, you will find it below. In javascript - my sample for .NET does > basically the same thing, with the same result, except that it also won't > drop the loopback-a call leg. > 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess > I'm not 100% sure what I'm doing.. :) What I want to be able to do is to > dial into a script, let the script dial another extension, and bridge them > together when the other party answers the call. I also need to take care of > call setup problems - if the other part doesn't respond, is unavailable or > busy in the phone - so I though this was the only way? If I use the > session.execute("bridge"..), will I be able to control the call if it > couldn't be connected? > > --- > > if (session.ready()) { > > session.answer(); > > new_session = new Session("loopback/5000", session); > new_session.waitForAnswer(); > > bridge(session, new_session); > > // Not sure if this is needed - I've tried with it both enabled and > disabled > session.hangup(); > new_session.hangup(); > } > > Peter > > > On 09-04-13 17.54, "Anthony Minessale" > wrote: > > 1) When you say latest, which rev does that mean? we change revs pretty > often. > 2) Do you have a minimal script that reproduces your issue. > 3) is there a reason you cannot just session.execute("bridge", dest); > instead of doing it manually (which is a process not for the faint at > heart)? > > > > On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > I have two problems that I haven't been able to solve. I've done the same > tests in both javascript, and in .NET. > > The two scripts are pretty simple, they just answer an incomming call, > creates a new session, wait for an answer on the second call leg, and then > bridge the two channels together. > > In both cases everything works just fine, but the audio is distorted. The > destination I'm calling is "loopback/5000" - the sample IVR application > included in FreeSWITCH. I first thought it was a codec issue, but even after > trying to switch to different codecs the problem was the same. It more > sounds like it's a timestamping issue - the voice is not distorted enough to > be a bad codec, but it reads way to fast (mayby twice the "normal" speed). > When doing a direct transfer() to the other destination this works just > fine, but I need to be able to have some extra logic to tell if the > destination is available or not. > > The second problem occurs only in .NET. After doing this sample there is as > loopback channel still hanging around. It seems like the call creates a > loopback-a and loopback-b, the loopback-b dissapears as it should (when the > call has been disconnected), but the other one stays there. When doing the > same in javascript this doesn't seem to occur. > > I'm using the latest SVN trunk, and my OS is Windows XP. > > I found bug FSCORE-349 in Jira, which seems to point in to the direction > that there might be a bug with the loopback channels in some cases, but I > could not find anything about the audio which plays too fast. > > Has anyone else experienced this? > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090413/32e37184/attachment-0002.html From berni at birkenwald.de Mon Apr 13 12:04:26 2009 From: berni at birkenwald.de (Bernhard Schmidt) Date: Mon, 13 Apr 2009 19:04:26 +0000 (UTC) Subject: [Freeswitch-users] Can't call registered user in internal-ipv6 profile References: <04D731AC-5A70-4515-8ABA-B432D18871FA@freeswitch.org> Message-ID: Brian West wrote: > user/ uses the dial-string in the domain to find the user on the > internal profile by default.. so to call someone registered via ipv6 > you'll need to put a dial-string param in the user to find them on the > internal-ipv6 profile. Oh okay, that works, thanks a lot. The default is from directory/default.xml, right? How does that point to the internal profile? Or is this choice hardcoded somewhere? Is there any pitfall replacing the default with it works, but I think there are a lot of things in Freeswitch I haven't fully understood yet. Regards, Bernhard >> probably a pretty easy problem, but I can't figure it out >> nevertheless. >> >> I'm still experimenting with FreeSwitch (SVN trunk, about two weeks >> old), mainly due to the IPv6 SIP support. I'm pretty much running the >> default configuration included in the sourcetree. Now I've hit the >> following problem: >> >> I have two phones registered, one Snom with extension 1000 on IPv4 >> (profile internal), one SIP Communicator with extension 1002 on IPv6 >> (profile internal-ipv6). I can call the Snom just fine (from the SIP >> Communicator as well as from outside or the CLI), but not the SIP >> Communicator. >> >> EXECUTE sofia/internal/1000 at obelix.oms16.birkenwald.de bridge(user/1002-BnVRy6f7ncgPVn3RC9QTCQ at public.gmane.org >> ) >> 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:1077 >> switch_ivr_originate() variable string 0 = [presence_id@?Hak?T??8????? >> ] >> 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 >> switch_ivr_originate() Cannot create outgoing channel of type >> [error] cause: [USER_NOT_REGISTERED] >> 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 >> switch_ivr_originate() Originate Resulted in Error Cause: 606 >> [USER_NOT_REGISTERED] >> 2009-04-13 18:23:45 [ERR] switch_ivr_originate.c:1486 >> switch_ivr_originate() Cannot create outgoing channel of type [user] >> cause: [USER_NOT_REGISTERED] >> 2009-04-13 18:23:45 [DEBUG] switch_ivr_originate.c:2084 >> switch_ivr_originate() Originate Resulted in Error Cause: 606 >> [USER_NOT_REGISTERED] >> 2009-04-13 18:23:45 [INFO] mod_dptools.c:2051 >> audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED >> >> It works when I explicitly specify the internal-ipv6 profile for the >> outgoing call like this: >> >> >> >> but that isn't really what I want. >> >> What are the quirks I need to add to have "user/@" >> search both profiles? I already set force-register-domain on the >> profile, but I don't think that is what I'm looking for. From diego.viola at gmail.com Mon Apr 13 22:36:09 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 01:36:09 -0400 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools Message-ID: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> Hi everyone, I have a question... I have this on my dialplan: What I want to do is play and read some digits and as soon as I get those digits, transfer to that extension... but this never happens, even if I terminate with a #. I do the same thing with Lua and it works with Lua, but I need it to work with play_and_get_digits from mod_dptools, because I plan to use this with event socket outbound, with an application which I'm currently working on. Any ideas? Thanks, Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/b9cfe5f8/attachment-0002.html From diego.viola at gmail.com Mon Apr 13 22:42:45 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 01:42:45 -0400 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools In-Reply-To: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> References: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> Message-ID: <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> It works if I use "read" and do this: But I need play_and_get_digits to work like that too, please. Diego On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote: > Hi everyone, > > I have a question... I have this on my dialplan: > > > > > > > > > What I want to do is play and read some digits and as soon as I get those > digits, transfer to that extension... but this never happens, even if I > terminate with a #. > > I do the same thing with Lua and it works with Lua, but I need it to work > with play_and_get_digits from mod_dptools, because I plan to use this with > event socket outbound, with an application which I'm currently working on. > > Any ideas? > > Thanks, > > Diego > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/ac771dc3/attachment-0002.html From diego.viola at gmail.com Mon Apr 13 22:53:39 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 01:53:39 -0400 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools In-Reply-To: <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> References: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> Message-ID: <86a32abc0904132253r57a4a9d3mf0f372473544276f@mail.gmail.com> I remember playAndGetDigits had a bug like this too. Anthony, please help me. On Tue, Apr 14, 2009 at 1:42 AM, Diego Viola wrote: > It works if I use "read" and do this: > > > > > > > > > > > But I need play_and_get_digits to work like that too, please. > > Diego > > > On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote: > >> Hi everyone, >> >> I have a question... I have this on my dialplan: >> >> >> >> >> >> >> >> >> What I want to do is play and read some digits and as soon as I get those >> digits, transfer to that extension... but this never happens, even if I >> terminate with a #. >> >> I do the same thing with Lua and it works with Lua, but I need it to work >> with play_and_get_digits from mod_dptools, because I plan to use this with >> event socket outbound, with an application which I'm currently working on. >> >> Any ideas? >> >> Thanks, >> >> Diego >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/332280d3/attachment-0002.html From moizchinoy at gmail.com Mon Apr 13 23:02:29 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Tue, 14 Apr 2009 10:02:29 +0400 Subject: [Freeswitch-users] Google Talk Integration... Message-ID: <29b888f80904132302k22e45594w997f70bd48be28e2@mail.gmail.com> Hi, I have tried google talk integration with FS and is working fine. Great Work! Is it possible to have multiple concurrent incoming calls on the same gmail account? -- Regards, Moiz Chinoy. From diego.viola at gmail.com Mon Apr 13 23:02:46 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 02:02:46 -0400 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools In-Reply-To: <86a32abc0904132253r57a4a9d3mf0f372473544276f@mail.gmail.com> References: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> <86a32abc0904132253r57a4a9d3mf0f372473544276f@mail.gmail.com> Message-ID: <86a32abc0904132302y41981bd9w11f6a4c9fac5b5c3@mail.gmail.com> Anthony, I just tried to print the variable with the log app, with read it prints, with play_and_get_digits doesn't. I'm using latest SVN rev: FreeSWITCH Version 1.0.trunk (13012M) Thanks, Diego On Tue, Apr 14, 2009 at 1:53 AM, Diego Viola wrote: > I remember playAndGetDigits had a bug like this too. > > Anthony, please help me. > > > On Tue, Apr 14, 2009 at 1:42 AM, Diego Viola wrote: > >> It works if I use "read" and do this: >> >> >> >> >> >> >> >> >> >> >> But I need play_and_get_digits to work like that too, please. >> >> Diego >> >> >> On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote: >> >>> Hi everyone, >>> >>> I have a question... I have this on my dialplan: >>> >>> >>> >>> >>> >>> >>> >>> >>> What I want to do is play and read some digits and as soon as I get those >>> digits, transfer to that extension... but this never happens, even if I >>> terminate with a #. >>> >>> I do the same thing with Lua and it works with Lua, but I need it to work >>> with play_and_get_digits from mod_dptools, because I plan to use this with >>> event socket outbound, with an application which I'm currently working on. >>> >>> Any ideas? >>> >>> Thanks, >>> >>> Diego >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/8d66df9e/attachment-0002.html From yudha2008 at gmail.com Tue Apr 14 00:24:48 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 14 Apr 2009 12:54:48 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> Message-ID: *Hi Brian West,* * I have installed the latest SVN Freeswitch trunk but still i get the same error. How can i over come this problem. 2009-04-14 12:44:26 [NOTICE] modjava.c:244 mod_java_load() Java Framework Loading... 2009-04-14 12:44:26 [ERR] modjava.c:133 load_config() Error loading /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so 2009-04-14 12:44:26 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** *2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_lua] 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:248 switch_loadable_module_process() Adding Application 'lua' 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'luarun' 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'lua' 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_say_en] 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:395 switch_loadable_module_process() Adding Say interface 'en' 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:120 switch_loadable_module_runtime() Starting runtime thread for CORE_SOFTTIMER_MODULE 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:120 switch_loadable_module_runtime() Starting runtime thread for mod_event_socket 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 switch_load_network_lists() Created ip list dl-candidates default (allow) 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 10.0.0.0/8 (deny) to list dl-candidates 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 172.16.0.0/12 (deny) to list dl-candidates 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 192.168.0.0/16 (deny) to list dl-candidates 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 switch_load_network_lists() Created ip list rfc1918 default (deny) 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 10.0.0.0/8 (allow) to list rfc1918 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 172.16.0.0/12 (allow) to list rfc1918 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 192.168.0.0/16 (allow) to list rfc1918 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 switch_load_network_lists() Created ip list lan default (allow) 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 192.168.42.0/24 (deny) to list lan 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 192.168.42.42/32 (allow) to list lan 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 switch_load_network_lists() Created ip list strict default (deny) 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 switch_load_network_lists() Adding 208.102.123.124/32 (allow) to list strict 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 switch_load_network_lists() Created ip list domains default (deny) 2009-04-14 12:44:26 [NOTICE] switch_core.c:965 switch_load_network_lists() Adding 192.0.2.0/24 (allow) [brian at 192.168.1.140] to list domains 2009-04-14 12:44:26 [CONSOLE] switch_core.c:1322 switch_core_init_and_modload() *FreeSWITCH Version 1.0.trunk (13013M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] Specify what is the error why i cant able to load Mod_java. -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/093cbce8/attachment-0002.html From solko at gcdf.pl Tue Apr 14 00:59:40 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 14 Apr 2009 09:59:40 +0200 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> Message-ID: <49E4426C.6010400@gcdf.pl> Baskar pisze: > *Hi Brian West,* > > > * I have installed the latest SVN Freeswitch trunk but still i get the > same error. How can i over come this problem. > > 2009-04-14 12:44:26 [NOTICE] modjava.c:244 mod_java_load() Java > Framework Loading... > 2009-04-14 12:44:26 [ERR] modjava.c:133 load_config() Error loading > /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so First of all do you have java installed in that path? If not edit configuration path. I do not use java mod for some time but I had no problem to load them, only reloading was a problem. > 2009-04-14 12:44:26 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_java.so > **Module load routine returned an error** > *2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:889 > switch_loadable_module_load_file() Successfully Loaded [mod_lua] > 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:248 > switch_loadable_module_process() Adding Application 'lua' > 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:270 > switch_loadable_module_process() Adding API Function 'luarun' > 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:270 > switch_loadable_module_process() Adding API Function 'lua' > 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:889 > switch_loadable_module_load_file() Successfully Loaded [mod_say_en] > 2009-04-14 12:44:26 [NOTICE] switch_loadable_module.c:395 > switch_loadable_module_process() Adding Say interface 'en' > 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:120 > switch_loadable_module_runtime() Starting runtime thread for > CORE_SOFTTIMER_MODULE > 2009-04-14 12:44:26 [CONSOLE] switch_loadable_module.c:120 > switch_loadable_module_runtime() Starting runtime thread for > mod_event_socket > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 > switch_load_network_lists() Created ip list dl-candidates default (allow) > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 10.0.0.0/8 (deny) > to list dl-candidates > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 172.16.0.0/12 > (deny) to list dl-candidates > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 192.168.0.0/16 > (deny) to list dl-candidates > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 > switch_load_network_lists() Created ip list rfc1918 default (deny) > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 10.0.0.0/8 > (allow) to list rfc1918 > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 172.16.0.0/12 > (allow) to list rfc1918 > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 192.168.0.0/16 > (allow) to list rfc1918 > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 > switch_load_network_lists() Created ip list lan default (allow) > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 192.168.42.0/24 > (deny) to list lan > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 192.168.42.42/32 > (allow) to list lan > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 > switch_load_network_lists() Created ip list strict default (deny) > 2009-04-14 12:44:26 [NOTICE] switch_core.c:981 > switch_load_network_lists() Adding 208.102.123.124/32 > (allow) to list strict > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:898 > switch_load_network_lists() Created ip list domains default (deny) > 2009-04-14 12:44:26 [NOTICE] switch_core.c:965 > switch_load_network_lists() Adding 192.0.2.0/24 > (allow) [brian at 192.168.1.140 ] to list domains > 2009-04-14 12:44:26 [CONSOLE] switch_core.c:1322 > switch_core_init_and_modload() > *FreeSWITCH Version 1.0.trunk (13013M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > > Specify what is the error why i cant able to load Mod_java. > > -- > Warm 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 From yudha2008 at gmail.com Tue Apr 14 02:37:07 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 14 Apr 2009 15:07:07 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: <49E4426C.6010400@gcdf.pl> References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> Message-ID: *Hi, I have installed latest java version jdk1.6.0 in this path /usr/java/jdk1.6.0_04/bin I have reconfigured FS ./configure --with-java=/usr/java/jdk1.6.0_04/bin, make, make install But when i run freeswitch in console i get this error. 2009-04-14 15:00:22 [ERR] modjava.c:133 load_config() Error loading /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so 2009-04-14 15:00:22 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** I cant able to load mod_java. Can any one specify what is the error. Thanks in advance. -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/7e9657e1/attachment-0002.html From solko at gcdf.pl Tue Apr 14 03:31:24 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 14 Apr 2009 12:31:24 +0200 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> Message-ID: <49E465FC.2080305@gcdf.pl> Baskar pisze: > *Hi, > > > I have installed latest java version jdk1.6.0 in this path > /usr/java/jdk1.6.0_04/bin > > I have reconfigured FS ./configure > --with-java=/usr/java/jdk1.6.0_04/bin, make, make install > > But when i run freeswitch in console i get this error. > Exactly, configure java module in config file conf/autoload_configs/java.conf.xml This is runtime configuration not build configuration. Set right javavm path. It must point to your libjvm.so file. > 2009-04-14 15:00:22 [ERR] modjava.c:133 load_config() Error loading > /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so > 2009-04-14 15:00:22 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_java.so > **Module load routine returned an error** > > > I cant able to load mod_java. Can any one specify what is the error. > Thanks in advance. > > -- > Warm 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 From kunalgrao at yahoo.co.in Mon Apr 13 22:09:32 2009 From: kunalgrao at yahoo.co.in (kunal rao) Date: Tue, 14 Apr 2009 10:39:32 +0530 (IST) Subject: [Freeswitch-users] running on visual studio and elaborative documentation Message-ID: <542787.91121.qm@web7601.mail.in.yahoo.com> Hi ? even I have downloaded FreeSWITCH and using MS Visual Studio 2008. It is building properly. I now want to configure it properly. Can you please give me directions and also some links for good detailed comparisons between Asterisk and FreeSWITCH and elaborative documentation for the same.. --- On Wed, 1/4/09, freeswitch-users-request at lists.freeswitch.org wrote: From: freeswitch-users-request at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 34, Issue 3 To: freeswitch-users at lists.freeswitch.org Date: Wednesday, 1 April, 2009, 7:29 PM 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: Originate and Conference (Peter P GMX) ???2. Compiler error for Windows XP (SP2) (Lewis Liu) ???3. Re: Call For Help: Janitor Projects (Anthony Minessale) ???4. Re: Compiler error for Windows XP (SP2) (Michael Jerris) ???5. Re: Call For Help: Janitor Projects (Raymond Chandler) ---------------------------------------------------------------------- Message: 1 Date: Wed, 01 Apr 2009 13:41:06 +0200 From: Peter P GMX Subject: Re: [Freeswitch-users] Originate and Conference To: freeswitch-users at lists.freeswitch.org Message-ID: <49D352D2.3070303 at gmx.net> Content-Type: text/plain; charset=ISO-8859-15 Hello Brian, I tried this (on trunk 12862), but still the same behaviour. It does not aks for a PIN. Neither when transfering directly to the conference nor by transfering to the dialplan extension where conference is handled. Best regards Peter Brian West schrieb: > Update again to svn trunk... btw 1.0.4 pre3 is out on > files.freeswitch.org > > /b > > On Mar 30, 2009, at 6:44 PM, Dan Le wrote: > >> I get similar behavior as Peter when trying to enter a locked >> conference. >> >> If I am just dialing from a phone to a conference (on a dialplan), it >> will properly lock me out. But if I do an originate command >> (originate sofia/internal/1001 &conference(3000)), it will drop me >> into the conference, even though it is suppose to be locked. >> >> I am using the released 1.0.3 tag. >> > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon!? http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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: 2 Date: Wed, 1 Apr 2009 18:41:44 +0800 From: Lewis Liu Subject: [Freeswitch-users] Compiler error for Windows XP (SP2) To: freeswitch-users at lists.freeswitch.org Message-ID: ??? <814e59990904010341y61b920c2h24bb8a50c8ae2f44 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We download FreeSWITCH from SVN Trunk and want to build it on MS Visual Studio 2008 with platform. But we got one error message when we build it. FreeSWITCH\libs\win32\sofia\debug\libsofia_sip_ua_static.lib is built fail. So many files are lost, such as mod_sofia.dll..... Could you help me me for this, Please?? Whether something is lost in MS Visual Studio 2008 ?? Thanks a lot!! Lewis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/b79932de/attachment-0001.html ------------------------------ Message: 3 Date: Wed, 1 Apr 2009 08:19:54 -0500 From: Anthony Minessale Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects To: freeswitch-users at lists.freeswitch.org Message-ID: ??? <191c3a030904010619i21ddd8fj81b020340907eb27 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" have a look. http://www.google.com/search?q=janitor+project The phrase has already been coined. If you look closely we have 2 different perspectives in this thread. mszlazak is seeking more of the higher level user documentation, the holy grail magic documentation that is like the hitchhikers guide to the galaxy or harry potter's marauder's map can tune into what you need to know or what you don't understand and magically adjusts.? This is normal, we have a lot of users like that.? The majority of users will treat us like they are buying the software from us and impose their expectations on us.? It's helpful to us, it lets us see things from their perspective. Seven is looking it at more from a developer's perspective, he's actually willing to take the time to add things to the wiki and he wants to understand how the code works.? This is a good thing too, there are far less people of this type in our community but they are crucial. Core developers document by explaining what they are doing to people like Seven or by putting a reminder in the commit notes which are later translated into the CHANGELOG for the releases.? Michael, the author of this thread has added countless pages of documentation to the wiki this way. It's easy to say the author should document everything.? There is close to 300,000 lines of code in just the src directory in the FreeSWITCH tree (that is all code we wrote not counting any of the depends libs or any other form of pre-existing code).? I personally wrote the majority of that code so, I really appricate it when the communiuty gives me a few minutes to take a break while they document it.? The best people to document the high level fuctionality? is not the author btw.? It's the first few people who use it.? Most likely they are developing a product from it and they intend to profit from it in one way or another and its a fair tradeoff to have the section of functionality explained to them in exchange from wikifying it from their perspective.? The perspective of the author will be dry and mechinacal where that first-time-user version of the documentation will make much more sense to future readers. When it comes to the low level documentation, the C functions, we also need someone to help us with that if they feel there is not enough.? We write code, we know how it works.? If other people cannot figure out how it works, they will ask us and in the end it will be doucmented.? About 5% or less of people in the community even have to look in the code for the core.? The whole point of the FreeSWITCH design is to push everything up to scripts, remote connections and dialplan logic to let people concentrate on good ideas instead of the evil logic necessary to properly engineer a telephony engine.? So I recommend anybody interested starts out making sure there is ample documentation for the embedded and external API for lua, js, perl, python, ESL etc.? Then anybody who really likes C code can start with the module API layer and then dig deeper into the core code and learn how it works and if the documentation is not enough, add some, we appriciate any help we can get. 2009/4/1 >? First off. I would not call it a "janitors project" since that may offend > some. A second problem is your notion that documentation is > "not-quite-as-important" a task as writing code. I'm think many would say > you have that backwards. There is nothing more effective in evolving > FreeSwitch than good documentation which helps further development and is an > important part of "customer service." Good customer service is then a part > of "sales and marketing." Much more often than not, It's sales and marketing > that is more important to making something a "real product"? than > engineering. "Build it and they will come" almost never works. > > Anyway, I think you need a new name for this project. > > >? -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org < > freeswitch-users at lists.freeswitch.org>; > freeswitch-dev at lists.freeswitch.org > Sent: Tue, 31 Mar 2009 5:10 pm > Subject: [Freeswitch-users] Call For Help: Janitor Projects > >? Dear FreeSWITCH Community: > > As you know, FreeSWITCH has been growing leaps and bounds and it's going to > keep growing as the word spreads. The core development team of Anthony, > Mike, and Brian are very appreciative of the community's help and > involvement in the project. Simply put: the community is awesome! > > Some have asked how they can help. Most of us are not software developers, > but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To > this end I've started a "janitor projects" wiki page: > > http://wiki.freeswitch.org/wiki/Janitor_Projects > > We say "janitor" projects because they are things that help keep the > project clean and organized, just like the janitor cleans an office, takes > out the trash, replaces the toilet paper, etc. These are valuable services > that we sometimes take for granted. However, I think we can all appreciate > that the FreeSWITCH project would be better served if the developers could > focus on writing code, fixing bugs, etc. and not on the easier, > not-quite-as-important janitorial tasks. To that end we are inviting all who > wish to volunteer to please visit the above wiki page and check out some of > the projects listed so far. Email me off list if you'd like to volunteer to > help. I'm maintaining a list of "janitors" and what they are helping with. > If you have ideas for other janitor projects then by all means email them to > me and we'll discuss them. > > Thanks again for being such a great community! > > -Michael S Collins > IRC: mercutioviz > > See you at ClueCon 2009!? http://www.cluecon.com > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > New Low Prices on Dell Laptops - Starting at $399 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/f8bc7440/attachment-0001.html ------------------------------ Message: 4 Date: Wed, 1 Apr 2009 09:55:10 -0400 From: Michael Jerris Subject: Re: [Freeswitch-users] Compiler error for Windows XP (SP2) To: freeswitch-users at lists.freeswitch.org Message-ID: <711C4390-ED0C-4A06-9AE8-652B24D0C776 at jerris.com> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes If you try to build just the sofia library, what are the first few? warnings and errors you get? Mike On Apr 1, 2009, at 6:41 AM, Lewis Liu wrote: > We download FreeSWITCH from SVN Trunk and want to build it on MS? > Visual Studio 2008 with platform. > But we got one error message when we build it. > FreeSWITCH\libs\win32\sofia\debug\libsofia_sip_ua_static.lib is? > built fail. > So many files are lost, such as mod_sofia.dll..... > Could you help me me for this, Please?? > Whether something is lost in MS Visual Studio 2008 ?? > Thanks a lot!! > Lewis ------------------------------ Message: 5 Date: Wed, 01 Apr 2009 09:59:15 -0400 From: Raymond Chandler Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects To: freeswitch-users at lists.freeswitch.org Message-ID: <49D37333.5080701 at freeswitch.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed seven wrote: > I know that. And I'd like to read code. Developers written great code? > and also plenty of comments(which is documentation) in code. However,? > there are sth. don't need to comment in code but should be available? > on wiki. E.g. I followed the svn commit log, and found? > sip_auth_username and sip_auth_password added, so I documented to the? > wiki. >??? That's the right attitude to have... now if there were more people doing that and less people complaining like little school girls, we could actually reach the next level in Open-Sourcetopia. -Ray ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 34, Issue 3 *********************************************** Check out the all-new Messenger 9.0! Go to http://in.messenger.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/fa0dcdeb/attachment-0002.html From mike at jerris.com Tue Apr 14 04:41:01 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 14 Apr 2009 07:41:01 -0400 Subject: [Freeswitch-users] running on visual studio and elaborative documentation In-Reply-To: <542787.91121.qm@web7601.mail.in.yahoo.com> References: <542787.91121.qm@web7601.mail.in.yahoo.com> Message-ID: http://wiki.freeswitch.org/wiki/Special:Search?search=asterisk&go=Go On Apr 14, 2009, at 1:09 AM, kunal rao wrote: > > Hi > > even I have downloaded FreeSWITCH and using MS Visual Studio 2008. > It is building properly. I now want to configure it properly. Can > you please give me directions and also some links for good detailed > comparisons between Asterisk and FreeSWITCH and elaborative > documentation for the same.. > --- On Wed, 1/4/09, freeswitch-users-request at lists.freeswitch.org > wrote: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/92ed86fd/attachment-0002.html From yudha2008 at gmail.com Tue Apr 14 04:54:04 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 14 Apr 2009 17:24:04 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: <49E465FC.2080305@gcdf.pl> References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> Message-ID: Hi, My Java.conf.xml *This is runtime configuration not build configuration. Set right javavm path. It must point to your libjvm.so file.* But still i have the same error . 2009-04-14 17:18:56 [ERR] modjava.c:133 load_config() Error loading /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so 2009-04-14 17:18:56 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** How to over come this problem. Some one help me to solve it. -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/6b5041c4/attachment-0002.html From yudha2008 at gmail.com Tue Apr 14 05:08:15 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 14 Apr 2009 17:38:15 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> Message-ID: *Hi, I have notice While reinstalling the Freeswitch i get this message While both make and make install commands making all mod_java Note: src/org/freeswitch/Launcher.java uses unchecked or unsafe operations. Note: Recompile with -Xlint:unchecked for details. I am using CentOS 5.2 with Latest Freeswitch trunk can any one guide to resolve the problem. Thanks in advance. -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/28d78b25/attachment-0002.html From solko at gcdf.pl Tue Apr 14 05:11:19 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 14 Apr 2009 14:11:19 +0200 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> Message-ID: <49E47D67.2080608@gcdf.pl> Baskar pisze: > Hi, > > > My Java.conf.xml > > > > > > > > > > > > *This is runtime configuration not build configuration. Set right javavm > path. > It must point to your libjvm.so file.* > > But still i have the same error . > You wrote that Java is installed in /usr/java/jdk1.6.0_04/bin so make this config file to point to it. Don't you see you are pointing wrong directory ? Find libjvm.so and put path to it in that tag. I did not gave you right configuration I just wanted to show you where to put this config value. Where is you libjvm.so located? > 2009-04-14 17:18:56 [ERR] modjava.c:133 load_config() Error loading > /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so > 2009-04-14 17:18:56 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_java.so > **Module load routine returned an error** > > How to over come this problem. Some one help me to solve it. > > -- > Warm 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 From yudha2008 at gmail.com Tue Apr 14 05:22:39 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 14 Apr 2009 17:52:39 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: <49E47D67.2080608@gcdf.pl> References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> <49E47D67.2080608@gcdf.pl> Message-ID: *Hi, I have not edited the java.conf.xml * *my libjvm.so file is loacted in this paths* * [localhost ~]# locate libjvm.so /usr/java/jdk1.6.0_04/jre/lib/i386/client/libjvm.so /usr/java/jdk1.6.0_04/jre/lib/i386/server/libjvm.so /usr/lib/gcj-4.1.1/libjvm.so /usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/i386/client/libjvm.so /usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/i386/server/libjvm.so* * In the above libjvm.so file which path should i specify it in the java.conf.xml Guide me where i am wrong. -- Warm Regards, N.Baskar* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/0972ed5f/attachment-0002.html From solko at gcdf.pl Tue Apr 14 05:23:16 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 14 Apr 2009 14:23:16 +0200 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> Message-ID: <49E48034.1040507@gcdf.pl> Baskar pisze: > *Hi, > > > I have notice While reinstalling the Freeswitch i get this message While > both make and make install commands > > making all mod_java > Note: src/org/freeswitch/Launcher.java uses unchecked or unsafe operations. > Note: Recompile with -Xlint:unchecked for details. > > I am using CentOS 5.2 with Latest Freeswitch trunk > Don't worry about that, this is just warning. New syntax for generics in Java, but this should not lead to any problems. > can any one guide to resolve the problem. Thanks in advance. This is not your main issue. Szymon Olko > > -- > Warm 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 From anthony.minessale at gmail.com Tue Apr 14 05:57:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Apr 2009 07:57:20 -0500 Subject: [Freeswitch-users] Google Talk Integration... In-Reply-To: <29b888f80904132302k22e45594w997f70bd48be28e2@mail.gmail.com> References: <29b888f80904132302k22e45594w997f70bd48be28e2@mail.gmail.com> Message-ID: <191c3a030904140557k4709388av45288f8c7fa58248@mail.gmail.com> yes, it should be. On Tue, Apr 14, 2009 at 1:02 AM, Moiz Chinoy wrote: > Hi, > > I have tried google talk integration with FS and is working fine. Great > Work! > Is it possible to have multiple concurrent incoming calls on the same > gmail account? > > -- > Regards, > Moiz Chinoy. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/ba3d62af/attachment-0002.html From solko at gcdf.pl Tue Apr 14 05:57:33 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 14 Apr 2009 14:57:33 +0200 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> <49E47D67.2080608@gcdf.pl> Message-ID: <49E4883D.1030305@gcdf.pl> Baskar pisze: > *Hi, > > I have not edited the java.conf.xml > * > *my libjvm.so file is loacted in this paths* > > * [localhost ~]# locate libjvm.so > /usr/java/jdk1.6.0_04/jre/lib/i386/client/libjvm.so > /usr/java/jdk1.6.0_04/jre/lib/i386/server/libjvm.so > /usr/lib/gcj-4.1.1/libjvm.so > /usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/i386/client/libjvm.so > /usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/i386/server/libjvm.so* > > * > In the above libjvm.so file which path should i specify it in the > java.conf.xml > > Guide me where i am wrong. > > Try this one first: /usr/java/jdk1.6.0_04/jre/lib/i386/client/libjvm.so if it will not work then try the one with server. This is sun Java implementation, don't sure if it will work with gcj, so don't try tem if you don't have to. > > -- > Warm 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 From anthony.minessale at gmail.com Tue Apr 14 06:15:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Apr 2009 08:15:16 -0500 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools In-Reply-To: <86a32abc0904132302y41981bd9w11f6a4c9fac5b5c3@mail.gmail.com> References: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> <86a32abc0904132253r57a4a9d3mf0f372473544276f@mail.gmail.com> <86a32abc0904132302y41981bd9w11f6a4c9fac5b5c3@mail.gmail.com> Message-ID: <191c3a030904140615p56d7f998pa5ce9a90cad62dcf@mail.gmail.com> try \d instead of \\d in your regex On Tue, Apr 14, 2009 at 1:02 AM, Diego Viola wrote: > Anthony, > > I just tried to print the variable with the log app, with read it prints, > with play_and_get_digits doesn't. > > I'm using latest SVN rev: > > FreeSWITCH Version 1.0.trunk (13012M) > > Thanks, > > Diego > > > On Tue, Apr 14, 2009 at 1:53 AM, Diego Viola wrote: > >> I remember playAndGetDigits had a bug like this too. >> >> Anthony, please help me. >> >> >> On Tue, Apr 14, 2009 at 1:42 AM, Diego Viola wrote: >> >>> It works if I use "read" and do this: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> But I need play_and_get_digits to work like that too, please. >>> >>> Diego >>> >>> >>> On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote: >>> >>>> Hi everyone, >>>> >>>> I have a question... I have this on my dialplan: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> What I want to do is play and read some digits and as soon as I get >>>> those digits, transfer to that extension... but this never happens, even if >>>> I terminate with a #. >>>> >>>> I do the same thing with Lua and it works with Lua, but I need it to >>>> work with play_and_get_digits from mod_dptools, because I plan to use this >>>> with event socket outbound, with an application which I'm currently working >>>> on. >>>> >>>> Any ideas? >>>> >>>> Thanks, >>>> >>>> Diego >>>> >>> >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/fc0876f3/attachment-0002.html From peter.olsson at visionutveckling.se Tue Apr 14 06:24:26 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 14 Apr 2009 15:24:26 +0200 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> Anthony, Yes, it seems to work correct now. I did a couple of test calls, and tha audio was good - thanks! Another question about this scenario... When doing a session.transfer("5000"), this will transfer the call directly into the dialplan without the use of loopback-channels. But that way it's not possible to do it in a controlled way. Shouldn't it be possible to do the same thing with a bridge? As soon as the call is bridged, it gets "rid of" unneccecary loopback channels, and connecting the two endpoints directly - cause by then it should be two "normal" endpoints talking? Regards, Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 13 april 2009 20:38 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... see how it works in latest trunk 13011 nontheless you can just say session.execute("bridge", "loopback/5000"); and get the same result without touching that other channel. when the call fails, you will have an originate_disposition variable in session you can check. On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson > wrote: 1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available.. 2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg. 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. :) What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected? --- if (session.ready()) { session.answer(); new_session = new Session("loopback/5000", session); new_session.waitForAnswer(); bridge(session, new_session); // Not sure if this is needed - I've tried with it both enabled and disabled session.hangup(); new_session.hangup(); } Peter On 09-04-13 17.54, "Anthony Minessale" > wrote: 1) When you say latest, which rev does that mean? we change revs pretty often. 2) Do you have a minimal script that reproduces your issue. 3) is there a reason you cannot just session.execute("bridge", dest); instead of doing it manually (which is a process not for the faint at heart)? On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson > wrote: I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 !DSPAM:49e3899632939315582408! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/c290958f/attachment-0002.html From gmaruzz at celliax.org Tue Apr 14 07:13:26 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 14 Apr 2009 16:13:26 +0200 Subject: [Freeswitch-users] skypiax Round Robin interface In-Reply-To: References: Message-ID: <7b197bef0904140713w69d7916au8f319ac37c138c11@mail.gmail.com> Hi Seven, thanks a lot for the patch and all the Skypiax action. I'm just back from Eastern vacations, let me clear the backlog and I'll be back on this in a couple days. Thanks again! gm Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Fri, Apr 10, 2009 at 8:38 PM, dujinfang wrote: > Hi, > > I made a patch, so skypiax is possible to do a RR hunt besides the > sequential interface ANY. > > Usage: > > originate skypiax/RR/other_skype_name > sk list > > http://jira.freeswitch.org/browse/MODENDP-211 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 gmaruzz at celliax.org Tue Apr 14 07:27:44 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 14 Apr 2009 16:27:44 +0200 Subject: [Freeswitch-users] Skypiax as a windows service In-Reply-To: <0D1E9E22CCAC4F98ADA9863FDFF7FB85@UVix> References: <0D1E9E22CCAC4F98ADA9863FDFF7FB85@UVix> Message-ID: <7b197bef0904140727o693e1e72jbf7470cc00afbb9d@mail.gmail.com> Hi UV, seems a difficult one this one. I have no much experience in RDP/terminal server. If there is no way to have (or fake) audio driver on RDP/terminal server apps, probably the Skype clients will not works (as you experienced). I'm sure, I've read it (:-) ), that Skype clients can be run on a Windows machine as services, without any user logged in. That is what I would explore in the future, just adding the How To to the wiki page. What you are experiencing seems to be different, seems to be specific to the RDP/terminal server usage. I'm I understanding you correctly (that this is specific to RDP)? Can you send me more info/hints? In parallel, I'm slowly working on a way to farm out the Skype clients from the FS servers, so to have the Skype clients running on different machines on the same LAN. I've a proof of concept working on Linux for one channel. You think this would solve your problems (having the Skype clients running on separate machines other than the machines running FS)? I'm just back from Easter vacations, please allow a couple days for the accumulated backlog ;-) Thanks a lot for taking the time to explore Skypiax and report this, gm Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Mon, Apr 13, 2009 at 1:32 PM, UV wrote: > Great work on Skypiax, Giovanni. > > > > We?ve tested it in our lab for sometime and it works very well. > > Unfortunately, when we tried deploying it on a production environment > (running Win2K3 server farm), we ran into a barrier: > > FS is running as terminal server console application (to be easily > maintained remotely by RDP) > This is because Win2K does not allow RDP to access system console (session > /userid 0) > Skype does not work on terminal server due to a well known disappearing > audio drivers problem, therefore it has to run either as a console or a > service (both on session 0). > FS can run well as a windows service > Skypiax seem to load as service, but it can?t find the skype client and exit > with the following error: > > 2009-04-13 20:54:14 [ERR] mod_skypiax.c:990 load_config() rev > 13006M[00000000|37???? ][ERRORA? 990? ][skype_user??? ][-1, 0, 0] Failed to > connect to a SKYPE API for interface_id=1, no SKYPE client running, please > (re)start Skype client. Skypiax exiting > > > > This situation prevents me to run skypiax in production. > > > > I understand from the wiki page that windows service is not done yet ? so I > presume this is a predicted outcome. > > > > Any idea when and if this is planned to be implemented? > > > > Keep up the good work! > > > > Cheers, > > UV > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Tue Apr 14 08:26:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Apr 2009 10:26:55 -0500 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> Message-ID: <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> yes, But if you plan is to bridge the call, the loopback channel is completely unnecessary. Be careful how much control you want =D getting a phone call up and running is more work than you think (see switch_ivr_originate.c) On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Anthony, > > > > Yes, it seems to work correct now. I did a couple of test calls, and tha > audio was good ? thanks! > > > > Another question about this scenario... > > > > When doing a session.transfer(?5000?), this will transfer the call directly > into the dialplan without the use of loopback-channels. But that way it?s > not possible to do it in a controlled way. Shouldn?t it be possible to do > the same thing with a bridge? As soon as the call is bridged, it gets ?rid > of? unneccecary loopback channels, and connecting the two endpoints directly > ? cause by then it should be two ?normal? endpoints talking? > > > > Regards, > > > > Peter > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 13 april 2009 20:38 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > see how it works in latest trunk 13011 > > nontheless you can just say > > session.execute("bridge", "loopback/5000"); > > and get the same result without touching that other channel. > > when the call fails, you will have an originate_disposition variable in > session you can check. > > > On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > 1. The latest trunk I've tried with is 13008. Since I'm not doing > anything for production yet (just testing/evaluating), so I tend to update > as soon as there is new version available.. > 2. Yep, you will find it below. In javascript - my sample for .NET does > basically the same thing, with the same result, except that it also won't > drop the loopback-a call leg. > 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess > I'm not 100% sure what I'm doing.. :) What I want to be able to do is to > dial into a script, let the script dial another extension, and bridge them > together when the other party answers the call. I also need to take care of > call setup problems - if the other part doesn't respond, is unavailable or > busy in the phone - so I though this was the only way? If I use the > session.execute("bridge"..), will I be able to control the call if it > couldn't be connected? > > --- > > if (session.ready()) { > > session.answer(); > > new_session = new Session("loopback/5000", session); > new_session.waitForAnswer(); > > bridge(session, new_session); > > // Not sure if this is needed - I've tried with it both enabled and > disabled > session.hangup(); > new_session.hangup(); > } > > Peter > > > > On 09-04-13 17.54, "Anthony Minessale" > wrote: > > 1) When you say latest, which rev does that mean? we change revs pretty > often. > 2) Do you have a minimal script that reproduces your issue. > 3) is there a reason you cannot just session.execute("bridge", dest); > instead of doing it manually (which is a process not for the faint at > heart)? > > > > On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > I have two problems that I haven't been able to solve. I've done the same > tests in both javascript, and in .NET. > > The two scripts are pretty simple, they just answer an incomming call, > creates a new session, wait for an answer on the second call leg, and then > bridge the two channels together. > > In both cases everything works just fine, but the audio is distorted. The > destination I'm calling is "loopback/5000" - the sample IVR application > included in FreeSWITCH. I first thought it was a codec issue, but even after > trying to switch to different codecs the problem was the same. It more > sounds like it's a timestamping issue - the voice is not distorted enough to > be a bad codec, but it reads way to fast (mayby twice the "normal" speed). > When doing a direct transfer() to the other destination this works just > fine, but I need to be able to have some extra logic to tell if the > destination is available or not. > > The second problem occurs only in .NET. After doing this sample there is as > loopback channel still hanging around. It seems like the call creates a > loopback-a and loopback-b, the loopback-b dissapears as it should (when the > call has been disconnected), but the other one stays there. When doing the > same in javascript this doesn't seem to occur. > > I'm using the latest SVN trunk, and my OS is Windows XP. > > I found bug FSCORE-349 in Jira, which seems to point in to the direction > that there might be a bug with the loopback channels in some cases, but I > could not find anything about the audio which plays too fast. > > Has anyone else experienced this? > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > !DSPAM:49e3899632939315582408! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/e38e0519/attachment-0002.html From peter.olsson at visionutveckling.se Tue Apr 14 08:59:13 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 14 Apr 2009 17:59:13 +0200 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE413034@cooper> Yes, I'm starting to realize that... :) but you to get everything right - if I want to bridge a call, using the dialplan, then the only way is to use loopback, right? If I don't want a loopback I'm able to bridge to the destination directly? //Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 14 april 2009 17:27 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... yes, But if you plan is to bridge the call, the loopback channel is completely unnecessary. Be careful how much control you want =D getting a phone call up and running is more work than you think (see switch_ivr_originate.c) On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson > wrote: Anthony, Yes, it seems to work correct now. I did a couple of test calls, and tha audio was good - thanks! Another question about this scenario... When doing a session.transfer("5000"), this will transfer the call directly into the dialplan without the use of loopback-channels. But that way it's not possible to do it in a controlled way. Shouldn't it be possible to do the same thing with a bridge? As soon as the call is bridged, it gets "rid of" unneccecary loopback channels, and connecting the two endpoints directly - cause by then it should be two "normal" endpoints talking? Regards, Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 13 april 2009 20:38 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... see how it works in latest trunk 13011 nontheless you can just say session.execute("bridge", "loopback/5000"); and get the same result without touching that other channel. when the call fails, you will have an originate_disposition variable in session you can check. On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson > wrote: 1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available.. 2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg. 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. :) What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected? --- if (session.ready()) { session.answer(); new_session = new Session("loopback/5000", session); new_session.waitForAnswer(); bridge(session, new_session); // Not sure if this is needed - I've tried with it both enabled and disabled session.hangup(); new_session.hangup(); } Peter On 09-04-13 17.54, "Anthony Minessale" > wrote: 1) When you say latest, which rev does that mean? we change revs pretty often. 2) Do you have a minimal script that reproduces your issue. 3) is there a reason you cannot just session.execute("bridge", dest); instead of doing it manually (which is a process not for the faint at heart)? On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson > wrote: I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 !DSPAM:49e4ade432931915915389! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/f882100e/attachment-0002.html From anthony.minessale at gmail.com Tue Apr 14 09:25:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Apr 2009 11:25:37 -0500 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE413034@cooper> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE413034@cooper> Message-ID: <191c3a030904140925x28e322bft389091547ca0d5d6@mail.gmail.com> The bridge application will let you bridge right to a destination on *another* box. If you want to connect to a local extension like 5000 you can use the transfer application or method. session.transfer("5000"); exit(); or session.execute("transfer", "5000"); exit(); On Tue, Apr 14, 2009 at 10:59 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Yes, I?m starting to realize that... :) but you to get everything right ? > if I want to bridge a call, using the dialplan, then the only way is to use > loopback, right? If I don?t want a loopback I?m able to bridge to the > destination directly? > > > > //Peter > > > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 14 april 2009 17:27 > > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > yes, > > But if you plan is to bridge the call, the loopback channel is completely > unnecessary. > Be careful how much control you want =D getting a phone call up and running > is more work > than you think (see switch_ivr_originate.c) > > On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > Anthony, > > > > Yes, it seems to work correct now. I did a couple of test calls, and tha > audio was good ? thanks! > > > > Another question about this scenario... > > > > When doing a session.transfer(?5000?), this will transfer the call directly > into the dialplan without the use of loopback-channels. But that way it?s > not possible to do it in a controlled way. Shouldn?t it be possible to do > the same thing with a bridge? As soon as the call is bridged, it gets ?rid > of? unneccecary loopback channels, and connecting the two endpoints directly > ? cause by then it should be two ?normal? endpoints talking? > > > > Regards, > > > > Peter > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 13 april 2009 20:38 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > see how it works in latest trunk 13011 > > nontheless you can just say > > session.execute("bridge", "loopback/5000"); > > and get the same result without touching that other channel. > > when the call fails, you will have an originate_disposition variable in > session you can check. > > On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > 1. The latest trunk I've tried with is 13008. Since I'm not doing > anything for production yet (just testing/evaluating), so I tend to update > as soon as there is new version available.. > 2. Yep, you will find it below. In javascript - my sample for .NET does > basically the same thing, with the same result, except that it also won't > drop the loopback-a call leg. > 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess > I'm not 100% sure what I'm doing.. :) What I want to be able to do is to > dial into a script, let the script dial another extension, and bridge them > together when the other party answers the call. I also need to take care of > call setup problems - if the other part doesn't respond, is unavailable or > busy in the phone - so I though this was the only way? If I use the > session.execute("bridge"..), will I be able to control the call if it > couldn't be connected? > > --- > > if (session.ready()) { > > session.answer(); > > new_session = new Session("loopback/5000", session); > new_session.waitForAnswer(); > > bridge(session, new_session); > > // Not sure if this is needed - I've tried with it both enabled and > disabled > session.hangup(); > new_session.hangup(); > } > > Peter > > > > On 09-04-13 17.54, "Anthony Minessale" > wrote: > > 1) When you say latest, which rev does that mean? we change revs pretty > often. > 2) Do you have a minimal script that reproduces your issue. > 3) is there a reason you cannot just session.execute("bridge", dest); > instead of doing it manually (which is a process not for the faint at > heart)? > > > > On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > I have two problems that I haven't been able to solve. I've done the same > tests in both javascript, and in .NET. > > The two scripts are pretty simple, they just answer an incomming call, > creates a new session, wait for an answer on the second call leg, and then > bridge the two channels together. > > In both cases everything works just fine, but the audio is distorted. The > destination I'm calling is "loopback/5000" - the sample IVR application > included in FreeSWITCH. I first thought it was a codec issue, but even after > trying to switch to different codecs the problem was the same. It more > sounds like it's a timestamping issue - the voice is not distorted enough to > be a bad codec, but it reads way to fast (mayby twice the "normal" speed). > When doing a direct transfer() to the other destination this works just > fine, but I need to be able to have some extra logic to tell if the > destination is available or not. > > The second problem occurs only in .NET. After doing this sample there is as > loopback channel still hanging around. It seems like the call creates a > loopback-a and loopback-b, the loopback-b dissapears as it should (when the > call has been disconnected), but the other one stays there. When doing the > same in javascript this doesn't seem to occur. > > I'm using the latest SVN trunk, and my OS is Windows XP. > > I found bug FSCORE-349 in Jira, which seems to point in to the direction > that there might be a bug with the loopback channels in some cases, but I > could not find anything about the audio which plays too fast. > > Has anyone else experienced this? > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > !DSPAM:49e4ade432931915915389! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/e0ce86cd/attachment-0002.html From kristian.kielhofner at gmail.com Tue Apr 14 09:51:37 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 14 Apr 2009 12:51:37 -0400 Subject: [Freeswitch-users] Adding Spanish support to say Message-ID: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> Hello everyone, I'm trying to add Spanish support to say. I'm using something like: in conf/lang/es which is included by freeswitch.conf: ..right after English. Yet I continue to get [ERR] switch_ivr.c:2014 switch_ivr_say() Invalid SAY Interface [es]! Whenever trying to use say: What am I missing? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Tue Apr 14 09:54:54 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 14 Apr 2009 12:54:54 -0400 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> Message-ID: <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> Replying to myself... I forgot to indicate my version! I am running trunk rev 12862 on CentOS 5 x86_64. On Tue, Apr 14, 2009 at 12:51 PM, Kristian Kielhofner wrote: > Hello everyone, > > ?I'm trying to add Spanish support to say. ?I'm using something like: > > > ? tts-engine="cepstral" tts-voice="callie"> > ? ? > ? ? > ? ? ? > ? > > > in conf/lang/es which is included by freeswitch.conf: > > > > ..right after English. ?Yet I continue to get > > [ERR] switch_ivr.c:2014 switch_ivr_say() Invalid SAY Interface [es]! > > ?Whenever trying to use say: > > > > ?What am I missing? > > Thanks! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Tue Apr 14 10:01:22 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Apr 2009 12:01:22 -0500 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> Message-ID: Nobody has written the es language files. Those would need to be written. /b On Apr 14, 2009, at 11:54 AM, Kristian Kielhofner wrote: > Replying to myself... I forgot to indicate my version! I am running > trunk rev 12862 on CentOS 5 x86_64. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/0a366926/attachment-0002.html From peter.olsson at visionutveckling.se Tue Apr 14 10:02:47 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 14 Apr 2009 19:02:47 +0200 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <191c3a030904140925x28e322bft389091547ca0d5d6@mail.gmail.com> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE413034@cooper> <191c3a030904140925x28e322bft389091547ca0d5d6@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE41303C@cooper> Allright - last question :) I'll try to be a little more specific. Lets say I whant to do the following; 1. Dial into FreeSWITCH, to some kind of application (javascript or whatever). 2. Answer that call, and let the user choose what to do; 1: record message, 2: transfer to XXX etc. The user presses 2. 3. I don't want to release the first call leg yet, since I need to be really sure that 2 is reachable (or else I will give the user choices again, with som kind of "the call could not be transferred"). So lets say I play some music for the user while trying to connect the call. 4. I originate another call - now I understand I have two choices, either I originate directly to a SIP phone (sofia/internal...), or I let the dialplan do the work - and if I want the dialplan to be the one to transfer the call somewhere (maybe to the same extension), I must use loopback - right? 5. If the new call answers, bridge the two calls, if it fails, start over again, after reading an error message. Whould this also be possible with transfer? If I understand everything right I loose control of the call, and won't be able to handle the failed transfer? Or is it possible to solve in a better way? What I guess I'd really want to do is to ask the dialplan "hey, I want to dial XXXX - give me the full sofia profile string" so I can originate the call directly, and I won't need a loopback. I could of course connect to the sofia string directly, but it would be nice to leave that kind of lookup logic to the dialplan. Thanks for staying with me - I hope you understand my problem :) Regards, Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 14 april 2009 18:26 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... The bridge application will let you bridge right to a destination on *another* box. If you want to connect to a local extension like 5000 you can use the transfer application or method. session.transfer("5000"); exit(); or session.execute("transfer", "5000"); exit(); On Tue, Apr 14, 2009 at 10:59 AM, Peter Olsson > wrote: Yes, I'm starting to realize that... :) but you to get everything right - if I want to bridge a call, using the dialplan, then the only way is to use loopback, right? If I don't want a loopback I'm able to bridge to the destination directly? //Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 14 april 2009 17:27 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... yes, But if you plan is to bridge the call, the loopback channel is completely unnecessary. Be careful how much control you want =D getting a phone call up and running is more work than you think (see switch_ivr_originate.c) On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson > wrote: Anthony, Yes, it seems to work correct now. I did a couple of test calls, and tha audio was good - thanks! Another question about this scenario... When doing a session.transfer("5000"), this will transfer the call directly into the dialplan without the use of loopback-channels. But that way it's not possible to do it in a controlled way. Shouldn't it be possible to do the same thing with a bridge? As soon as the call is bridged, it gets "rid of" unneccecary loopback channels, and connecting the two endpoints directly - cause by then it should be two "normal" endpoints talking? Regards, Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 13 april 2009 20:38 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts... see how it works in latest trunk 13011 nontheless you can just say session.execute("bridge", "loopback/5000"); and get the same result without touching that other channel. when the call fails, you will have an originate_disposition variable in session you can check. On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson > wrote: 1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available.. 2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg. 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. :) What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected? --- if (session.ready()) { session.answer(); new_session = new Session("loopback/5000", session); new_session.waitForAnswer(); bridge(session, new_session); // Not sure if this is needed - I've tried with it both enabled and disabled session.hangup(); new_session.hangup(); } Peter On 09-04-13 17.54, "Anthony Minessale" > wrote: 1) When you say latest, which rev does that mean? we change revs pretty often. 2) Do you have a minimal script that reproduces your issue. 3) is there a reason you cannot just session.execute("bridge", dest); instead of doing it manually (which is a process not for the faint at heart)? On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson > wrote: I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 !DSPAM:49e4bcb132932104520616! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/c1d31444/attachment-0002.html From anthony.minessale at gmail.com Tue Apr 14 10:21:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Apr 2009 12:21:19 -0500 Subject: [Freeswitch-users] Use of loopback channels and bridge() in scripts... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4AAE41303C@cooper> References: <191c3a030904130854s6df22924t123b3ffdbcb45452@mail.gmail.com> <191c3a030904131137y85502f7hdc1e1aae424e7454@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE412FFA@cooper> <191c3a030904140826n7cdf139dt9858c8598108cae3@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE413034@cooper> <191c3a030904140925x28e322bft389091547ca0d5d6@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4AAE41303C@cooper> Message-ID: <191c3a030904141021mf4d873bvf361135dd677ea0f@mail.gmail.com> typically you would use transfer to the dest then in the dialplan for XXXX you would set hangup_after_bridge=true try to call the phone transfer back to your ivr you can use channel variables to keep track of state. On Tue, Apr 14, 2009 at 12:02 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Allright ? last question :) I?ll try to be a little more specific. Lets > say I whant to do the following; > > > > 1. Dial into FreeSWITCH, to some kind of application (javascript or > whatever). > > 2. Answer that call, and let the user choose what to do; 1: record > message, 2: transfer to XXX etc. The user presses 2. > > 3. I don?t want to release the first call leg yet, since I need to > be really sure that 2 is reachable (or else I will give the user choices > again, with som kind of ?the call could not be transferred?). So lets say I > play some music for the user while trying to connect the call. > > 4. I originate another call ? now I understand I have two choices, > either I originate directly to a SIP phone (sofia/internal...), or I let the > dialplan do the work ? and if I want the dialplan to be the one to transfer > the call somewhere (maybe to the same extension), I must use loopback ? > right? > > 5. If the new call answers, bridge the two calls, if it fails, start > over again, after reading an error message. > > > > Whould this also be possible with transfer? If I understand everything > right I loose control of the call, and won?t be able to handle the failed > transfer? Or is it possible to solve in a better way? > > > > What I guess I?d really want to do is to ask the dialplan ?hey, I want to > dial XXXX ? give me the full sofia profile string? so I can originate the > call directly, and I won?t need a loopback. I could of course connect to the > sofia string directly, but it would be nice to leave that kind of lookup > logic to the dialplan. > > > > Thanks for staying with me ? I hope you understand my problem :) > > > > Regards, > > > > Peter > > > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 14 april 2009 18:26 > > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > The bridge application will let you bridge right to a destination on > *another* box. > If you want to connect to a local extension like 5000 you can use the > transfer application or method. > > session.transfer("5000"); > exit(); > > or > > session.execute("transfer", "5000"); > exit(); > > > On Tue, Apr 14, 2009 at 10:59 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > Yes, I?m starting to realize that... :) but you to get everything right ? > if I want to bridge a call, using the dialplan, then the only way is to use > loopback, right? If I don?t want a loopback I?m able to bridge to the > destination directly? > > > > //Peter > > > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 14 april 2009 17:27 > > > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > yes, > > But if you plan is to bridge the call, the loopback channel is completely > unnecessary. > Be careful how much control you want =D getting a phone call up and running > is more work > than you think (see switch_ivr_originate.c) > > On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > Anthony, > > > > Yes, it seems to work correct now. I did a couple of test calls, and tha > audio was good ? thanks! > > > > Another question about this scenario... > > > > When doing a session.transfer(?5000?), this will transfer the call directly > into the dialplan without the use of loopback-channels. But that way it?s > not possible to do it in a controlled way. Shouldn?t it be possible to do > the same thing with a bridge? As soon as the call is bridged, it gets ?rid > of? unneccecary loopback channels, and connecting the two endpoints directly > ? cause by then it should be two ?normal? endpoints talking? > > > > Regards, > > > > Peter > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale > *Skickat:* den 13 april 2009 20:38 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Use of loopback channels and bridge() in > scripts... > > > > see how it works in latest trunk 13011 > > nontheless you can just say > > session.execute("bridge", "loopback/5000"); > > and get the same result without touching that other channel. > > when the call fails, you will have an originate_disposition variable in > session you can check. > > On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > > 1. The latest trunk I've tried with is 13008. Since I'm not doing > anything for production yet (just testing/evaluating), so I tend to update > as soon as there is new version available.. > 2. Yep, you will find it below. In javascript - my sample for .NET does > basically the same thing, with the same result, except that it also won't > drop the loopback-a call leg. > 3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess > I'm not 100% sure what I'm doing.. :) What I want to be able to do is to > dial into a script, let the script dial another extension, and bridge them > together when the other party answers the call. I also need to take care of > call setup problems - if the other part doesn't respond, is unavailable or > busy in the phone - so I though this was the only way? If I use the > session.execute("bridge"..), will I be able to control the call if it > couldn't be connected? > > --- > > if (session.ready()) { > > session.answer(); > > new_session = new Session("loopback/5000", session); > new_session.waitForAnswer(); > > bridge(session, new_session); > > // Not sure if this is needed - I've tried with it both enabled and > disabled > session.hangup(); > new_session.hangup(); > } > > Peter > > > > On 09-04-13 17.54, "Anthony Minessale" > wrote: > > 1) When you say latest, which rev does that mean? we change revs pretty > often. > 2) Do you have a minimal script that reproduces your issue. > 3) is there a reason you cannot just session.execute("bridge", dest); > instead of doing it manually (which is a process not for the faint at > heart)? > > > > On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > I have two problems that I haven't been able to solve. I've done the same > tests in both javascript, and in .NET. > > The two scripts are pretty simple, they just answer an incomming call, > creates a new session, wait for an answer on the second call leg, and then > bridge the two channels together. > > In both cases everything works just fine, but the audio is distorted. The > destination I'm calling is "loopback/5000" - the sample IVR application > included in FreeSWITCH. I first thought it was a codec issue, but even after > trying to switch to different codecs the problem was the same. It more > sounds like it's a timestamping issue - the voice is not distorted enough to > be a bad codec, but it reads way to fast (mayby twice the "normal" speed). > When doing a direct transfer() to the other destination this works just > fine, but I need to be able to have some extra logic to tell if the > destination is available or not. > > The second problem occurs only in .NET. After doing this sample there is as > loopback channel still hanging around. It seems like the call creates a > loopback-a and loopback-b, the loopback-b dissapears as it should (when the > call has been disconnected), but the other one stays there. When doing the > same in javascript this doesn't seem to occur. > > I'm using the latest SVN trunk, and my OS is Windows XP. > > I found bug FSCORE-349 in Jira, which seems to point in to the direction > that there might be a bug with the loopback channels in some cases, but I > could not find anything about the audio which plays too fast. > > Has anyone else experienced this? > > Regards, > > Peter Olsson > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > !DSPAM:49e4bcb132932104520616! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/107b32fe/attachment-0002.html From msc at freeswitch.org Tue Apr 14 10:25:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Apr 2009 10:25:34 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre4 Now Available Message-ID: <87f2f3b90904141025h4df3320ai60d9710ab6449f26@mail.gmail.com> The FreeSWITCH team is pleased to announce the immediate availability of version 1.0.4pre4. Details are available here: http://www.freeswitch.org/node/173 All are encouraged to upgrade as soon as possible. Thanks to everyone for their feedback, ideas, and bug reports. Please keep them coming. -Michael http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/16f9e5b0/attachment-0002.html From msc at freeswitch.org Tue Apr 14 10:37:27 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Apr 2009 10:37:27 -0700 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> Message-ID: <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> KK, Do you have someone who knows Spanish and who can translate? If not I will whip up some volunteers from the FS community. Thanks, MC On Tue, Apr 14, 2009 at 10:01 AM, Brian West wrote: > Nobody has written the es language files. Those would need to be written. > /b > > On Apr 14, 2009, at 11:54 AM, Kristian Kielhofner wrote: > > Replying to myself... I forgot to indicate my version! I am running > trunk rev 12862 on CentOS 5 x86_64. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/b4a2497f/attachment-0002.html From jmesquita at gmail.com Tue Apr 14 10:48:38 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 14 Apr 2009 14:48:38 -0300 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> Message-ID: <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> I know spanish and I would translate it no problem. MC, get in touch with me off-list so we can handle that. I can also translate to portuguese-brazil. jmesquita On Apr 14, 2009, at 2:37 PM, Michael Collins wrote: > KK, > Do you have someone who knows Spanish and who can translate? If not > I will whip up some volunteers from the FS community. > > Thanks, > MC > > On Tue, Apr 14, 2009 at 10:01 AM, Brian West > wrote: > Nobody has written the es language files. Those would need to be > written. > > /b > > On Apr 14, 2009, at 11:54 AM, Kristian Kielhofner wrote: > >> Replying to myself... I forgot to indicate my version! I am running >> trunk rev 12862 on CentOS 5 x86_64. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/990f33ae/attachment-0002.html From brian at freeswitch.org Tue Apr 14 10:57:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Apr 2009 12:57:44 -0500 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> Message-ID: <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> This also requires you to write all the phrase macros for voicemail, ivr and other things in the demo in lang/en/ /b On Apr 14, 2009, at 12:48 PM, Jo?o Mesquita wrote: > I know spanish and I would translate it no problem. MC, get in touch > with me off-list so we can handle that. > > I can also translate to portuguese-brazil. > > jmesquita Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/9979ea58/attachment-0002.html From kristian.kielhofner at gmail.com Tue Apr 14 11:07:14 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 14 Apr 2009 14:07:14 -0400 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> Message-ID: <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> Brian, For my application I just need to be able to say a string of numbers - Caller ID, etc. Other than the files used there is no syntax or grammar difference (in Spanish) when compared to English. I should just be able to drop the files in. I'll have a problem when I need to handle IVR, voicemail, and other more complex issues but this will solve my immediate needs. For now I'm just trying to figure out how to get language "es" recognized by say... On Tue, Apr 14, 2009 at 1:57 PM, Brian West wrote: > This also requires you to write all the phrase macros for voicemail, ivr and > other things in the demo in lang/en/ > /b > On Apr 14, 2009, at 12:48 PM, Jo?o Mesquita wrote: > > I know spanish and I would translate it no problem. MC, get in touch with me > off-list so we can handle that. > I can also translate to portuguese-brazil. > jmesquita > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From msc at freeswitch.org Tue Apr 14 11:12:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Apr 2009 11:12:20 -0700 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> Message-ID: <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> Cool. We've had several volunteers start translating the phrase files into Spanish and Brazilian Portugese. We'll keep you posted when we have the Spanish one ready. FYI, I committed a stub phrase_es.xml file but it hasn't been translated yet except for the first twenty digits. However, there aren't any audio files associated with it yet... -MC On Tue, Apr 14, 2009 at 11:07 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Brian, > > For my application I just need to be able to say a string of numbers > - Caller ID, etc. > > Other than the files used there is no syntax or grammar difference > (in Spanish) when compared to English. I should just be able to drop > the files in. > > I'll have a problem when I need to handle IVR, voicemail, and other > more complex issues but this will solve my immediate needs. > > For now I'm just trying to figure out how to get language "es" > recognized by say... > > On Tue, Apr 14, 2009 at 1:57 PM, Brian West wrote: > > This also requires you to write all the phrase macros for voicemail, ivr > and > > other things in the demo in lang/en/ > > /b > > On Apr 14, 2009, at 12:48 PM, Jo?o Mesquita wrote: > > > > I know spanish and I would translate it no problem. MC, get in touch with > me > > off-list so we can handle that. > > I can also translate to portuguese-brazil. > > jmesquita > > > > Brian West > > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/553c3d06/attachment-0002.html From diego.viola at gmail.com Tue Apr 14 11:48:40 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 14:48:40 -0400 Subject: [Freeswitch-users] Help with play_and_get_digits from mod_dptools In-Reply-To: <191c3a030904140615p56d7f998pa5ce9a90cad62dcf@mail.gmail.com> References: <86a32abc0904132236m61b22d33t776910db7f1f6d79@mail.gmail.com> <86a32abc0904132242r181333bcl481bf67480f6cdfe@mail.gmail.com> <86a32abc0904132253r57a4a9d3mf0f372473544276f@mail.gmail.com> <86a32abc0904132302y41981bd9w11f6a4c9fac5b5c3@mail.gmail.com> <191c3a030904140615p56d7f998pa5ce9a90cad62dcf@mail.gmail.com> Message-ID: <86a32abc0904141148l33454b54m533cb556bd8f6515@mail.gmail.com> That works, thanks Anthm, you're the man. Diego On Tue, Apr 14, 2009 at 9:15 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try \d instead of \\d in your regex > > On Tue, Apr 14, 2009 at 1:02 AM, Diego Viola wrote: > >> Anthony, >> >> I just tried to print the variable with the log app, with read it prints, >> with play_and_get_digits doesn't. >> >> I'm using latest SVN rev: >> >> FreeSWITCH Version 1.0.trunk (13012M) >> >> Thanks, >> >> Diego >> >> >> On Tue, Apr 14, 2009 at 1:53 AM, Diego Viola wrote: >> >>> I remember playAndGetDigits had a bug like this too. >>> >>> Anthony, please help me. >>> >>> >>> On Tue, Apr 14, 2009 at 1:42 AM, Diego Viola wrote: >>> >>>> It works if I use "read" and do this: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> But I need play_and_get_digits to work like that too, please. >>>> >>>> Diego >>>> >>>> >>>> On Tue, Apr 14, 2009 at 1:36 AM, Diego Viola wrote: >>>> >>>>> Hi everyone, >>>>> >>>>> I have a question... I have this on my dialplan: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> What I want to do is play and read some digits and as soon as I get >>>>> those digits, transfer to that extension... but this never happens, even if >>>>> I terminate with a #. >>>>> >>>>> I do the same thing with Lua and it works with Lua, but I need it to >>>>> work with play_and_get_digits from mod_dptools, because I plan to use this >>>>> with event socket outbound, with an application which I'm currently working >>>>> on. >>>>> >>>>> Any ideas? >>>>> >>>>> Thanks, >>>>> >>>>> Diego >>>>> >>>> >>>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/73eed9e9/attachment-0002.html From diego.viola at gmail.com Tue Apr 14 11:56:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 14 Apr 2009 14:56:02 -0400 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> Message-ID: <86a32abc0904141156x4bd04908pa0a83854d4617f56@mail.gmail.com> Hey guys, If you need some Spanish help count with my help also. Diego On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins wrote: > Cool. We've had several volunteers start translating the phrase files into > Spanish and Brazilian Portugese. We'll keep you posted when we have the > Spanish one ready. FYI, I committed a stub phrase_es.xml file but it hasn't > been translated yet except for the first twenty digits. However, there > aren't any audio files associated with it yet... > > -MC > > > On Tue, Apr 14, 2009 at 11:07 AM, Kristian Kielhofner < > kristian.kielhofner at gmail.com> wrote: > >> Brian, >> >> For my application I just need to be able to say a string of numbers >> - Caller ID, etc. >> >> Other than the files used there is no syntax or grammar difference >> (in Spanish) when compared to English. I should just be able to drop >> the files in. >> >> I'll have a problem when I need to handle IVR, voicemail, and other >> more complex issues but this will solve my immediate needs. >> >> For now I'm just trying to figure out how to get language "es" >> recognized by say... >> >> On Tue, Apr 14, 2009 at 1:57 PM, Brian West wrote: >> > This also requires you to write all the phrase macros for voicemail, ivr >> and >> > other things in the demo in lang/en/ >> > /b >> > On Apr 14, 2009, at 12:48 PM, Jo?o Mesquita wrote: >> > >> > I know spanish and I would translate it no problem. MC, get in touch >> with me >> > off-list so we can handle that. >> > I can also translate to portuguese-brazil. >> > jmesquita >> > >> > Brian West >> > brian at freeswitch.org >> > -- Meet us at ClueCon! http://www.cluecon.com >> > >> > >> > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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://blog.krisk.org >> http://www.submityoursip.com >> http://www.astlinux.org >> http://www.star2star.com >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/30398a21/attachment-0002.html From nicolas at medularis.com Tue Apr 14 12:21:02 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 14 Apr 2009 15:21:02 -0400 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <86a32abc0904141156x4bd04908pa0a83854d4617f56@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> <86a32abc0904141156x4bd04908pa0a83854d4617f56@mail.gmail.com> Message-ID: <1b46b4e80904141221h16cfa78ci807f8541f66b31b1@mail.gmail.com> I'm a native spanish speaker, I can help too! Nicol?s Brenner On Tue, Apr 14, 2009 at 2:56 PM, Diego Viola wrote: > Hey guys, > > If you need some Spanish help count with my help also. > > Diego > > > On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins wrote: > >> Cool. We've had several volunteers start translating the phrase files into >> Spanish and Brazilian Portugese. We'll keep you posted when we have the >> Spanish one ready. FYI, I committed a stub phrase_es.xml file but it hasn't >> been translated yet except for the first twenty digits. However, there >> aren't any audio files associated with it yet... >> >> -MC >> >> >> On Tue, Apr 14, 2009 at 11:07 AM, Kristian Kielhofner < >> kristian.kielhofner at gmail.com> wrote: >> >>> Brian, >>> >>> For my application I just need to be able to say a string of numbers >>> - Caller ID, etc. >>> >>> Other than the files used there is no syntax or grammar difference >>> (in Spanish) when compared to English. I should just be able to drop >>> the files in. >>> >>> I'll have a problem when I need to handle IVR, voicemail, and other >>> more complex issues but this will solve my immediate needs. >>> >>> For now I'm just trying to figure out how to get language "es" >>> recognized by say... >>> >>> On Tue, Apr 14, 2009 at 1:57 PM, Brian West >>> wrote: >>> > This also requires you to write all the phrase macros for voicemail, >>> ivr and >>> > other things in the demo in lang/en/ >>> > /b >>> > On Apr 14, 2009, at 12:48 PM, Jo?o Mesquita wrote: >>> > >>> > I know spanish and I would translate it no problem. MC, get in touch >>> with me >>> > off-list so we can handle that. >>> > I can also translate to portuguese-brazil. >>> > jmesquita >>> > >>> > Brian West >>> > brian at freeswitch.org >>> > -- Meet us at ClueCon! http://www.cluecon.com >>> > >>> > >>> > >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at 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://blog.krisk.org >>> http://www.submityoursip.com >>> http://www.astlinux.org >>> http://www.star2star.com >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/12169928/attachment-0002.html From kristian.kielhofner at gmail.com Tue Apr 14 12:58:54 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 14 Apr 2009 15:58:54 -0400 Subject: [Freeswitch-users] Adding Spanish support to say In-Reply-To: <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> References: <2d9149cd0904140951t5d8abcacl8805646e5ad159a2@mail.gmail.com> <2d9149cd0904140954m23d4e4a9kc66b1f6db16f6485@mail.gmail.com> <87f2f3b90904141037w117aa679m81b4dab6ea4ce89c@mail.gmail.com> <737272AB-7803-43D9-A256-4629B531DDB6@gmail.com> <34E47744-AC7A-499C-BEA6-3EE6AAF3FD29@freeswitch.org> <2d9149cd0904141107s29c05e4ci1ced86f7a0a5240e@mail.gmail.com> <87f2f3b90904141112r764961ah57cc5e4187b41c36@mail.gmail.com> Message-ID: <2d9149cd0904141258i24988b8iadb4cf87e8ce073f@mail.gmail.com> On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins wrote: > Cool. We've had several volunteers start translating the phrase files into > Spanish and Brazilian Portugese. We'll keep you posted when we have the > Spanish one ready. FYI, I committed a stub phrase_es.xml file but it hasn't > been translated yet except for the first twenty digits. However, there > aren't any audio files associated with it yet... > > -MC > Is there anything I can do with this file now? I can't seem to find the relationship between any of the phrase files in that directory and my running configuration. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From mfedyk at mikefedyk.com Tue Apr 14 20:59:54 2009 From: mfedyk at mikefedyk.com (Mike Fedyk) Date: Tue, 14 Apr 2009 20:59:54 -0700 Subject: [Freeswitch-users] Recommended tools for creating/extending a sip test suite? Message-ID: <93cdabd20904142059h37091d46lc40e571f21553f91@mail.gmail.com> Hi all, I'm looking for suggestions on which open source tools to use for creating (or extending if there is already a project for this) a sip test suite. I have already heard of sipp, but I want to know what others are using and how they go about this before starting from scratch myself. Some things I'd like to do: - Dialplan/ voice menu/provider/did testing: Call number, press 1, expect to receive call on another extension. (kinda like expect) - Load testing Basically I want to be able to automate how a human may interact with my installation to reproduce bugs and make sure they don't come back. That way I can make sure my changes (wherever they may be in my stack, dialplan, freeswitch, openser/kamailio/opensips, etc.). Any pointers and/or tips will be much appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/5ee6118b/attachment-0002.html From mitul at enterux.com Wed Apr 15 00:22:50 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 15 Apr 2009 03:22:50 -0400 Subject: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ? Message-ID: <27149.1239780170@enterux.com> BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; }Hello there, In my previous encounter with FreeSwitch, I had found that Bret had posted on the Mailing List somewhere about availability of the entire FreeSwitch Wiki Documentation on a single PDF, this is useful coz at the offset apart from Wiki there is no other offline media to learn it. Is the same PDF available looking at the growth of Wiki pages and the updation. I look forward to hear from you guys, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions, The Enterprise Linux Company (TM), www.enterux.com +91-9820332422 ------------------------- Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090415/20372f3f/attachment-0002.html From stevecrozz at gmail.com Tue Apr 14 22:39:31 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Tue, 14 Apr 2009 22:39:31 -0700 Subject: [Freeswitch-users] Recommended tools for creating/extending a sip test suite? In-Reply-To: <93cdabd20904142059h37091d46lc40e571f21553f91@mail.gmail.com> References: <93cdabd20904142059h37091d46lc40e571f21553f91@mail.gmail.com> Message-ID: <11990ade0904142239j757fca09jac61837c313c714c@mail.gmail.com> It seems to me like the freeswitch platform itself would be a good place to start. I haven't thoroughly thought this out, but maybe you could write a test library using mod_ designed to do human-like things such as issuing dtmf tones, pausing, speaking, etc. You could even run test scripts using the event socket (api commands) and test the results by subscribing to related events. I'd love to hear about what you come up with. --Stephen On Tue, Apr 14, 2009 at 8:59 PM, Mike Fedyk wrote: > Hi all, > > I'm looking for suggestions on which open source tools to use for creating > (or extending if there is already a project for this) a sip test suite. > > I have already heard of sipp, but I want to know what others are using and > how they go about this before starting from scratch myself. > > Some things I'd like to do: > - Dialplan/ voice menu/provider/did testing: Call number, press 1, expect > to receive call on another extension. (kinda like expect) > - Load testing > > Basically I want to be able to automate how a human may interact with my > installation to reproduce bugs and make sure they don't come back. That way > I can make sure my changes (wherever they may be in my stack, dialplan, > freeswitch, openser/kamailio/opensips, etc.). > > Any pointers and/or tips will be much appreciated. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/5c14e419/attachment-0002.html From yudha2008 at gmail.com Wed Apr 15 00:15:52 2009 From: yudha2008 at gmail.com (Baskar) Date: Wed, 15 Apr 2009 12:45:52 +0530 Subject: [Freeswitch-users] Mod_java loading error In-Reply-To: <49E4883D.1030305@gcdf.pl> References: <0D35638F-28AC-4C6A-92C7-66E251F48DEC@freeswitch.org> <49E4426C.6010400@gcdf.pl> <49E465FC.2080305@gcdf.pl> <49E47D67.2080608@gcdf.pl> <49E4883D.1030305@gcdf.pl> Message-ID: *Hi, Now i can able to load the mod_java in the freeswitch console. After that i have followed these method to run the PhoneTest.java * *1) verified my classpath in the java.conf.xml: ] However, none of the files in conf have a tag called . All files are conforming xml. I can't seem to find what's changed. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/b8b33f1d/attachment-0002.html From brian at freeswitch.org Wed Apr 29 08:32:55 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 29 Apr 2009 10:32:55 -0500 Subject: [Freeswitch-users] Very confusing startup error In-Reply-To: <98a86adf0904290821s7f7314e9laef0029115016441@mail.gmail.com> References: <98a86adf0904290821s7f7314e9laef0029115016441@mail.gmail.com> Message-ID: <9B4982CF-1E5B-4297-87EE-F76CD2AE4321@freeswitch.org> The first is an error that is unrelated to the second error. Check out freeswitch.xml.fsxml line 2423 you'll have an extra line there. /b On Apr 29, 2009, at 10:21 AM, Gerry Hull wrote: > All of a sudden I'm getting this startup error when I start > FreeSwitch: > > C:\DVLP\FreeSwitch>freeswitch > Error including C:\DVLP\FreeSwitch\conf\autoload_configs\.. > \sip_profiles\internal/*.xml (Invalid argument) > Cannot Initialize [[error near line 2423]: unexpected closing tag context>] > > However, none of the files in conf have a tag called . > All files are conforming xml. I can't seem to find what's changed. > > Any ideas? > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/fd782cc3/attachment-0002.html From gk at exram.de Wed Apr 29 08:36:13 2009 From: gk at exram.de (Guido Kuth) Date: Wed, 29 Apr 2009 15:36:13 +0000 Subject: [Freeswitch-users] Very confusing startup error Message-ID: At least your dialplan should have a tag named . See default dialplan ! Original Message processed by David.InfoCenter Subject: [Freeswitch-users] Very confusing startup error (29-Apr-2009 17:26) From: Gerry Hull To: gk at exram.de All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitch>freeswitch Error including C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml (Invalid argument) Cannot Initialize [[error near line 2423]: unexpected closing tag ] However, none of the files in conf have a tag called . All files are conforming xml. I can't seem to find what's changed. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/ae5db11f/attachment-0002.html From gerry at pstn2.net Wed Apr 29 08:54:14 2009 From: gerry at pstn2.net (Gerry Hull) Date: Wed, 29 Apr 2009 11:54:14 -0400 Subject: [Freeswitch-users] Very confusing startup error In-Reply-To: References: Message-ID: <98a86adf0904290854v44ca7491qcff74454b3599b46@mail.gmail.com> Thanks Guys! I could not find my problem -- but you pointed me in the correct direction. I had a mismatched tag in my public.xml in the dialpan. So, is freeswitch.xml.fsxml a logged representation of the complete config file in memory? On Wed, Apr 29, 2009 at 11:36 AM, Guido Kuth wrote: > At least your dialplan should have a tag named . See default > dialplan ! > > > > Original Message > * processed by David.InfoCenter* > Subject: > [Freeswitch-users] Very confusing startup error (29-Apr-2009 17:26) > From: > Gerry Hull > To: > gk at exram.de > > All of a sudden I'm getting this startup error when I start FreeSwitch: > > C:\DVLP\FreeSwitch>freeswitch > Error including > C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml > (Invalid argument) > Cannot Initialize [[error near line 2423]: unexpected closing tag > ] > > However, none of the files in conf have a tag called . All > files are conforming xml. I can't seem to find what's changed. > > Any ideas? > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/68be6acc/attachment-0002.html From q.edward at gmail.com Wed Apr 29 09:12:16 2009 From: q.edward at gmail.com (Edward Q.) Date: Wed, 29 Apr 2009 12:12:16 -0400 Subject: [Freeswitch-users] HELP 3-way network access In-Reply-To: <89313a90904290902p2146f19co86695c3576073b5d@mail.gmail.com> References: <89313a90904290902p2146f19co86695c3576073b5d@mail.gmail.com> Message-ID: <89313a90904290912n171c867evd1368c26cfc01ade@mail.gmail.com> Hi guys .. I need your help please... I am trying to setup an FS box. It has to be like a 3 way thing since i reside in one network - my FS machine resides on another network - and my provider (gateway) resides on another network. I am going to try to be specific as much as i can. I did a Quick and Dirty install. Here is my testing servers info. Hardware Intel Dual Core 2.6 GHZ Real memory 3.56 GB total, 267.71 MB used Hard Drives 1 SATA 250GB MotherBoard Biostar P4M900-M4 Motherboard - VIA P4M900, Socket 478, MicroATX Software Operating system CentOS Linux 5.2 Kernel and CPU Linux 2.6.18-92.1.22.el5 on i686 Apache 2.2.3 MySQL 5.0.45 SSH OpenSSH 4.3 freeswitch at internal> version FreeSWITCH Version 1.0.trunk (13181M) Ok the FS testing server resides on xxx.9.10.xxx. The gateway resides on xxx.9.9.xxx. And my computer resides on 75.74.xxx.xxx (My computer has X-lite) installed. When i create the SIP profile on X-Lite in my computer and tell X-Lite to register on xxx.9.10.xxx It says discovering network ... Initializing... Registering ... And then it shows up your Your username is: 1000 (looks like it is registered). Now when i try to dial 5000 to listen at least to the IVR demo i get ... The person you are calling is unavailable please try again ... message. and shows on the top of the username ... Call failed: Request Timeout (message) And on the fs_cli console it shows this .. freeswitch at internal> 2009-04-29 11:46:05 [DEBUG] sofia.c:4242 sofia_handle_sip_i_invite() IP 75.74.xxx.xxx Rejected by acl "domains". Falling back to Digest auth. I am a total noob on this. I replaced my original acl.conf.xml with this... I shutdown FS and then restart FS with the -nc option. And Still the same thing. My gateway is a CANTATA switch which does not require authentication. I am trying to generate a call from my 75.74.xxx.xxx using X-Lite to a PSTN phone on the outside using the CANTATA switch on xxx.9.9.xxx through my FS box on xxx.9.10.xxx But as for now I can't even get the IVR to work for now ... Since i don't know anything about FS i would like to know what am i doing wrong.. And what files have to be either created or updated to do this. Thanks to everyone for all the help Edward -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/46a04212/attachment-0002.html From gallo at mctelefonia.com Wed Apr 29 09:18:04 2009 From: gallo at mctelefonia.com (Antonio Gallo) Date: Wed, 29 Apr 2009 18:18:04 +0200 Subject: [Freeswitch-users] is there something like "flash operator panel" for freeswitch? In-Reply-To: <191c3a030904290537p6f460642ucedd85b3856374e2@mail.gmail.com> References: <49F5B799.80809@mctelefonia.com> <191c3a030904270720j78438eecr6ae3418358d7e727@mail.gmail.com> <49F5C4A9.7000506@mctelefonia.com> <2D0785C2-9CC1-46E7-9EEE-55E49D63CB2C@freeswitch.org> <49F5C7D5.2090003@mctelefonia.com> <49F7FB59.1090608@mctelefonia.com> <191c3a030904290537p6f460642ucedd85b3856374e2@mail.gmail.com> Message-ID: <49F87DBC.7010203@mctelefonia.com> ok i did some test today using the yesterday's trunk with a gxp2010 and a snom360 both with 2 LEDS monitoring each other and themselves. Configuration: gxp2010 user: 1000 led1: 1000 led2: 1001 snom360 user: 1001 led1: 1000 led2: 1001 Problem with both phones: - when a phone reboot and it subscribe it does not get notified of the current status of the subscribed phones i.e. if gxp is on the phone the snom led1 is off/unlit i.e. if snom is on the phone the gxp led2 is off/unlit Problem with GXP only: - both subscribe LED stop working after a 1 or 2 calls until the gxp re-subscribe or re-register To skip using LED on phones is there is something like "flash operator panel" to display telephone status? Thanks in advance, Antonio (AGX) From brian at freeswitch.org Wed Apr 29 09:19:02 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 29 Apr 2009 11:19:02 -0500 Subject: [Freeswitch-users] HELP 3-way network access In-Reply-To: <89313a90904290912n171c867evd1368c26cfc01ade@mail.gmail.com> References: <89313a90904290902p2146f19co86695c3576073b5d@mail.gmail.com> <89313a90904290912n171c867evd1368c26cfc01ade@mail.gmail.com> Message-ID: <665E076F-A898-47E5-920C-2BBDA75C19AB@freeswitch.org> Now you need to open up the sofia profile in sip_profile/internal.xml and apply the test1 acl instead of the "domains" acl. /b On Apr 29, 2009, at 11:12 AM, Edward Q. wrote: > freeswitch at internal> 2009-04-29 11:46:05 [DEBUG] sofia.c:4242 > sofia_handle_sip_i_invite() IP 75.74.xxx.xxx Rejected by acl > "domains". Falling back to Digest auth. > > I am a total noob on this. I replaced my original acl.conf.xml with > this... > > > > > > > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/17b5afa3/attachment-0002.html From msc at freeswitch.org Wed Apr 29 09:37:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Apr 2009 09:37:19 -0700 Subject: [Freeswitch-users] Very confusing startup error In-Reply-To: <98a86adf0904290854v44ca7491qcff74454b3599b46@mail.gmail.com> References: <98a86adf0904290854v44ca7491qcff74454b3599b46@mail.gmail.com> Message-ID: <87f2f3b90904290937j51e51a8ew1dfc43b1e0d1675f@mail.gmail.com> On Wed, Apr 29, 2009 at 8:54 AM, Gerry Hull wrote: > Thanks Guys! > > I could not find my problem -- but you pointed me in the correct > direction. I had a mismatched tag in my public.xml in the dialpan. > > So, is freeswitch.xml.fsxml a logged representation of the complete config > file in memory? > > Affirmative. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/99679635/attachment-0002.html From Prometheus001 at gmx.net Wed Apr 29 10:03:45 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 29 Apr 2009 19:03:45 +0200 Subject: [Freeswitch-users] First Linux soft phone running TLS/SRTP on Linux (Zoipe Bizz) Message-ID: <49F88871.8000409@gmx.net> After 6 months of discussions with Attractel, today we finally got a new version of Zoiper Bizz, which works with TLS and SRTP (previous versions only supported TLS). I have added the info, how to set it up, in the wiki http://wiki.freeswitch.org/wiki/Interop_List#Zoiper_Bizz_2.10_and_TLS.2FSRTP We've been searching for a long time to have a working secure VoIP client under Linux. So far Zoiper seems to be the only VoIP soft phone capable of managing TLS/SRTP with Freeswitch under Linux. BTW: The free version does not support encryption. The Zoiper "Bizz" version does, but is not for free. Best regards Peter From brian at freeswitch.org Wed Apr 29 10:12:31 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 29 Apr 2009 12:12:31 -0500 Subject: [Freeswitch-users] First Linux soft phone running TLS/SRTP on Linux (Zoipe Bizz) In-Reply-To: <49F88871.8000409@gmx.net> References: <49F88871.8000409@gmx.net> Message-ID: <66B1E03E-0C70-4E91-8DA2-5067E5A6DA17@freeswitch.org> Lets not forget FreeSWITCH is a soft phone also that could do TLS and SRTP too :) /b On Apr 29, 2009, at 12:03 PM, Peter P GMX wrote: > So far Zoiper seems to be the only VoIP soft phone capable of managing > TLS/SRTP with Freeswitch under Linux. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/da43426f/attachment-0002.html From gk at exram.de Wed Apr 29 10:21:54 2009 From: gk at exram.de (Guido Kuth) Date: Wed, 29 Apr 2009 17:21:54 +0000 Subject: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL Message-ID: I have a problem I am trying to solve for several days now. I have FS 1.3.0 installed. I have the default configuration except that I have edited event_socket.conf to match my configuration. I have two computers with x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from 1000 and my esl app answers the call plays back a greeting and after that sends a record_session command and a start_dtmf command. Now I send the bridge command with sofia/internal/1001 at ip-address. The x-lite 1001 rings and I can take the call the two can talk to each other and both are able to end the call by hanging up the phone, but there is no reaction on any dtmf tone except when I press * and 1-3, cause this is defined by bind-meta-app in default dialplan. What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Any help or suggestion is welcome. Thanks in advance...Guido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/545d60e1/attachment-0002.html From brian at freeswitch.org Wed Apr 29 10:30:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 29 Apr 2009 12:30:21 -0500 Subject: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL In-Reply-To: References: Message-ID: If you subscribe to the event you will receive one on every DTMF press if FreeSWITCH gets it... if you happen to be getting them via inband you won't receive an event unless you enable the inband detection app. http://wiki.freeswitch.org/wiki/Event_list#DTMF http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf I also highly recommend you update to SVN trunk. /b On Apr 29, 2009, at 12:21 PM, Guido Kuth wrote: > What I need is that I get an Event on DTMF Entry on the bridged > call. Please I have to resolve this, cause this is the reason why I > came from Asterisk to FreeSwitch. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/de5dc2ca/attachment-0002.html From anthony.minessale at gmail.com Wed Apr 29 10:39:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 29 Apr 2009 12:39:46 -0500 Subject: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL In-Reply-To: References: Message-ID: <191c3a030904291039u1501410bgbd15e422b9c3b916@mail.gmail.com> set the async flag on the socket app call that triggers your ESL connection On Wed, Apr 29, 2009 at 12:21 PM, Guido Kuth wrote: > I have a problem I am trying to solve for several days now. I have FS > 1.3.0 installed. I have the default configuration except that I have edited > event_socket.conf to match my configuration. I have two computers with > x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from > 1000 and my esl app answers the call plays back a greeting and after that > sends a record_session command and a start_dtmf command. > > Now I send the bridge command with sofia/internal/1001 at ip-address. The > x-lite 1001 rings and I can take the call the two can talk to each other and > both are able to end the call by hanging up the phone, but there is no > reaction on any dtmf tone except when I press * and 1-3, cause this is > defined by bind-meta-app in default dialplan. > > What I need is that I get an Event on DTMF Entry on the bridged call. > Please I have to resolve this, cause this is the reason why I came from > Asterisk to FreeSwitch. > > Any help or suggestion is welcome. > > Thanks in advance...Guido > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/c9daf86d/attachment-0002.html From gk at exram.de Wed Apr 29 10:48:32 2009 From: gk at exram.de (Guido Kuth) Date: Wed, 29 Apr 2009 17:48:32 +0000 Subject: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL Message-ID: First thanks for your reply. I have subscribed to all Events, so this can't be the mistake. I sent start_dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to "info"? I think setting it to "rfc2833" would not be very meanigfull. I will try to update to svn trunk tomorrow. Again thanks for first help...Guido Original Message processed by David.InfoCenter Subject: Re: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:34) From: Brian West To: gk at exram.de If you subscribe to the event you will receive one on every DTMF press if FreeSWITCH gets it... if you happen to be getting them via inband you won't receive an event unless you enable the inband detection app. http://wiki.freeswitch.org/wiki/Event_list#DTMF http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf I also highly recommend you update to SVN trunk. /b On Apr 29, 2009, at 12:21 PM, Guido Kuth wrote: What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/add9514a/attachment-0002.html From gk at exram.de Wed Apr 29 10:52:21 2009 From: gk at exram.de (Guido Kuth) Date: Wed, 29 Apr 2009 17:52:21 +0000 Subject: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL Message-ID: Hello Anthony, sorry, but I forgot to tell you that I have an inbound ESL connection not an outbound one. So I connect to FS and then wait for Events. I know that I can set async flag in outbound socket, but is this also possible for inbound socket, and when, is it the same as in outbound socket behind the IP-Address? Thank you very much...Guido Original Message processed by David.InfoCenter Subject: Re: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:47) From: Anthony Minessale To: gk at exram.de set the async flag on the socket app call that triggers your ESL connection On Wed, Apr 29, 2009 at 12:21 PM, Guido Kuth wrote: I have a problem I am trying to solve for several days now. I have FS 1.3.0 installed. I have the default configuration except that I have edited event_socket.conf to match my configuration. I have two computers with x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from 1000 and my esl app answers the call plays back a greeting and after that sends a record_session command and a start_dtmf command. Now I send the bridge command with sofia/internal/1001 at ip-address. The x-lite 1001 rings and I can take the call the two can talk to each other and both are able to end the call by hanging up the phone, but there is no reaction on any dtmf tone except when I press * and 1-3, cause this is defined by bind-meta-app in default dialplan. What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Any help or suggestion is welcome. Thanks in advance...Guido _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/7a6e0dab/attachment-0002.html From brian at freeswitch.org Wed Apr 29 10:54:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 29 Apr 2009 12:54:39 -0500 Subject: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL In-Reply-To: References: Message-ID: Well the best option is to NOT use inband at all if possible. And use RFC2833 which eyebeam/xlite support as do most providers out there... You do not HAVE to start_dtmf on sip channels unless they only send the DTMF inband. set the dtmf-type back to rfc2833 and restart FS. /b On Apr 29, 2009, at 12:48 PM, Guido Kuth wrote: > First thanks for your reply. > > I have subscribed to all Events, so this can't be the mistake. I > sent start_dtmf app to FreeSwitch in caller channel and the wiki > says that you have to do this on sip channels to enable inband dtmf. > I checked sofia.conf and I have found that param dtmf-type is > commented out. Would it be helpful to set this to "info"? I think > setting it to "rfc2833" would not be very meanigfull. > > I will try to update to svn trunk tomorrow. > > Again thanks for first help...Guido Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/39d7dfed/attachment-0002.html From paul.degt at gmail.com Wed Apr 29 12:15:28 2009 From: paul.degt at gmail.com (paul.degt) Date: Wed, 29 Apr 2009 15:15:28 -0400 Subject: [Freeswitch-users] Phones become unreachable after some time Message-ID: <49F8A750.7030906@gmail.com> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds to Mysql DB for SIP registrations, presence etc. I noticed that after some time probably >30 min. phones which have been registered but without making calls become unreachable. Meaning that any call to such extension gets forwarded to VM as if it was offline, until I reload such phone. I did try to make the phones to register every 5 min. but it does not help. I also see valid registration information in sip_registrations table. X-Lite has r-port and keep alive settings on. Would appreciate any hints on what can be the issue here. From gk at exram.de Wed Apr 29 12:20:42 2009 From: gk at exram.de (Guido Kuth) Date: Wed, 29 Apr 2009 19:20:42 +0000 Subject: [Freeswitch-users] Re-2: Re: Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL Message-ID: <0002D0FE.49F8C4AA@192.168.49.2> Thank you again Brian. The reason why I want to test with inband dtmf is that in the real environment FS will be behind a conventional ISDN PBX which will work as a gateway to the ISDN Network. So I do not know if the PBX will do something like a translation between DTMF Tones to rfc and backwars. If you have experience with this any help will be very welcome....Guido Original Message processed by David InfoCenter Subject: Re: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:59) From: Brian West To: freeswitch-users at lists.freeswitch.org Well the best option is to NOT use inband at all if possible. And use RFC2833 which eyebeam/xlite support as do most providers out there... You do not HAVE to start_dtmf on sip channels unless they only send the DTMF inband. set the dtmf-type back to rfc2833 and restart FS. /b On Apr 29, 2009, at 12:48 PM, Guido Kuth wrote: First thanks for your reply. I have subscribed to all Events, so this can't be the mistake. I sent start_ dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to "info"? I think setting it to "rfc2833" would not be very meanigfull. I will try to update to svn trunk tomorrow. Again thanks for first help...Guido Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/67e16cae/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Apr 29 12:41:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 29 Apr 2009 20:41:53 +0100 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: <49F8A750.7030906@gmail.com> References: <49F8A750.7030906@gmail.com> Message-ID: Do the phones and FS have a firewall between them? If so, sounds like the pin hole in the fw is being closed. Alot only stay open for 4 mins Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of paul.degt Sent: 29 April 2009 20:15 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Phones become unreachable after some time I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds to Mysql DB for SIP registrations, presence etc. I noticed that after some time probably >30 min. phones which have been registered but without making calls become unreachable. Meaning that any call to such extension gets forwarded to VM as if it was offline, until I reload such phone. I did try to make the phones to register every 5 min. but it does not help. I also see valid registration information in sip_registrations table. X-Lite has r-port and keep alive settings on. Would appreciate any hints on what can be the issue here. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 paul.degt at gmail.com Wed Apr 29 12:50:12 2009 From: paul.degt at gmail.com (paul.degt) Date: Wed, 29 Apr 2009 15:50:12 -0400 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: References: <49F8A750.7030906@gmail.com> Message-ID: <49F8AF74.5040004@gmail.com> They do, but all necessary ports for FS are open. If that is fw issue, are there ways to fight with it? Nik Middleton wrote: > Do the phones and FS have a firewall between them? If so, sounds like > the pin hole in the fw is being closed. Alot only stay open for 4 mins > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 29 April 2009 20:15 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Phones become unreachable after some time > > I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds > to Mysql DB for SIP registrations, presence etc. > I noticed that after some time probably >30 min. phones which have been > registered but without making calls become unreachable. Meaning that any > > call to such extension gets forwarded to VM as if it was offline, until > I reload such phone. > I did try to make the phones to register every 5 min. but it does not > help. I also see valid registration information in sip_registrations > table. X-Lite has r-port and keep alive settings on. > Would appreciate any hints on what can be the issue here. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 nik.middleton at noblesolutions.co.uk Wed Apr 29 14:07:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 29 Apr 2009 22:07:33 +0100 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: <49F8AF74.5040004@gmail.com> References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> Message-ID: Don't know where the setting is in FS, but force them to register every 120 seconds and see if that helps Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of paul.degt Sent: 29 April 2009 20:50 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Phones become unreachable after some time They do, but all necessary ports for FS are open. If that is fw issue, are there ways to fight with it? Nik Middleton wrote: > Do the phones and FS have a firewall between them? If so, sounds like > the pin hole in the fw is being closed. Alot only stay open for 4 mins > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 29 April 2009 20:15 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Phones become unreachable after some time > > I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds > to Mysql DB for SIP registrations, presence etc. > I noticed that after some time probably >30 min. phones which have been > registered but without making calls become unreachable. Meaning that any > > call to such extension gets forwarded to VM as if it was offline, until > I reload such phone. > I did try to make the phones to register every 5 min. but it does not > help. I also see valid registration information in sip_registrations > table. X-Lite has r-port and keep alive settings on. > Would appreciate any hints on what can be the issue here. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 mikael at bjerkeland.com Wed Apr 29 14:13:05 2009 From: mikael at bjerkeland.com (Mikael Bjerkeland) Date: Wed, 29 Apr 2009 23:13:05 +0200 Subject: [Freeswitch-users] is there something like "flash operator panel" for freeswitch? In-Reply-To: <49F87DBC.7010203@mctelefonia.com> References: <49F5B799.80809@mctelefonia.com> <191c3a030904270720j78438eecr6ae3418358d7e727@mail.gmail.com> <49F5C4A9.7000506@mctelefonia.com> <2D0785C2-9CC1-46E7-9EEE-55E49D63CB2C@freeswitch.org> <49F5C7D5.2090003@mctelefonia.com> <49F7FB59.1090608@mctelefonia.com> <191c3a030904290537p6f460642ucedd85b3856374e2@mail.gmail.com> <49F87DBC.7010203@mctelefonia.com> Message-ID: None that I know of, but it should be fairly simple to create FOP for FS with the event socket. 2009/4/29 Antonio Gallo > ok i did some test today using the yesterday's trunk with a gxp2010 and > a snom360 both with 2 LEDS monitoring each other and themselves. > > Configuration: > gxp2010 user: 1000 led1: 1000 led2: 1001 > snom360 user: 1001 led1: 1000 led2: 1001 > > Problem with both phones: > - when a phone reboot and it subscribe it does not get notified of the > current status of the subscribed phones > i.e. if gxp is on the phone the snom led1 is off/unlit > i.e. if snom is on the phone the gxp led2 is off/unlit > > Problem with GXP only: > - both subscribe LED stop working after a 1 or 2 calls until the gxp > re-subscribe or re-register > > > To skip using LED on phones is there is something like "flash operator > panel" to display > telephone status? > > Thanks in advance, > Antonio (AGX) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/34f6e48a/attachment-0002.html From msc at freeswitch.org Wed Apr 29 14:20:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Apr 2009 14:20:21 -0700 Subject: [Freeswitch-users] is there something like "flash operator panel" for freeswitch? In-Reply-To: References: <49F5B799.80809@mctelefonia.com> <191c3a030904270720j78438eecr6ae3418358d7e727@mail.gmail.com> <49F5C4A9.7000506@mctelefonia.com> <2D0785C2-9CC1-46E7-9EEE-55E49D63CB2C@freeswitch.org> <49F5C7D5.2090003@mctelefonia.com> <49F7FB59.1090608@mctelefonia.com> <191c3a030904290537p6f460642ucedd85b3856374e2@mail.gmail.com> <49F87DBC.7010203@mctelefonia.com> Message-ID: <87f2f3b90904291420r21af3c71tdbeb2df79baca489@mail.gmail.com> On Wed, Apr 29, 2009 at 2:13 PM, Mikael Bjerkeland wrote: > None that I know of, but it should be fairly simple to create FOP for FS > with the event socket. > The "fairly simple" part is actually doing it. The really hard part is finding the time/energy/inclination to do it... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/a42beb80/attachment-0002.html From anthony.minessale at gmail.com Wed Apr 29 15:22:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 29 Apr 2009 17:22:26 -0500 Subject: [Freeswitch-users] is there something like "flash operator panel" for freeswitch? In-Reply-To: <49F87DBC.7010203@mctelefonia.com> References: <49F5B799.80809@mctelefonia.com> <191c3a030904270720j78438eecr6ae3418358d7e727@mail.gmail.com> <49F5C4A9.7000506@mctelefonia.com> <2D0785C2-9CC1-46E7-9EEE-55E49D63CB2C@freeswitch.org> <49F5C7D5.2090003@mctelefonia.com> <49F7FB59.1090608@mctelefonia.com> <191c3a030904290537p6f460642ucedd85b3856374e2@mail.gmail.com> <49F87DBC.7010203@mctelefonia.com> Message-ID: <191c3a030904291522u62e66b99wdc7ce91feb76470b@mail.gmail.com> edit autoload_configs/sofia.conf.xml in add then you will see all the sql stmts etc and you can debug your issue On Wed, Apr 29, 2009 at 11:18 AM, Antonio Gallo wrote: > ok i did some test today using the yesterday's trunk with a gxp2010 and > a snom360 both with 2 LEDS monitoring each other and themselves. > > Configuration: > gxp2010 user: 1000 led1: 1000 led2: 1001 > snom360 user: 1001 led1: 1000 led2: 1001 > > Problem with both phones: > - when a phone reboot and it subscribe it does not get notified of the > current status of the subscribed phones > i.e. if gxp is on the phone the snom led1 is off/unlit > i.e. if snom is on the phone the gxp led2 is off/unlit > > Problem with GXP only: > - both subscribe LED stop working after a 1 or 2 calls until the gxp > re-subscribe or re-register > > > To skip using LED on phones is there is something like "flash operator > panel" to display > telephone status? > > Thanks in advance, > Antonio (AGX) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090429/474b58f4/attachment-0002.html From cervajs at fpf.slu.cz Thu Apr 30 00:23:47 2009 From: cervajs at fpf.slu.cz (marek cervenka) Date: Thu, 30 Apr 2009 09:23:47 +0200 (CEST) Subject: [Freeswitch-users] First Linux soft phone running TLS/SRTP on Linux (Zoipe Bizz) In-Reply-To: <49F88871.8000409@gmx.net> References: <49F88871.8000409@gmx.net> Message-ID: > After 6 months of discussions with Attractel, today we finally got a new > version of Zoiper Bizz, which works with TLS and SRTP (previous versions > only supported TLS). > I have added the info, how to set it up, in the wiki > http://wiki.freeswitch.org/wiki/Interop_List#Zoiper_Bizz_2.10_and_TLS.2FSRTP > > We've been searching for a long time to have a working secure VoIP > client under Linux. > So far Zoiper seems to be the only VoIP soft phone capable of managing > TLS/SRTP with Freeswitch under Linux. QuteCom have TLS/SRTP on its roadmap http://trac.qutecom.org/roadmap --------------------------------------- Marek Cervenka ======================================= From rossmck at mac.com Thu Apr 30 02:46:44 2009 From: rossmck at mac.com (Ross McKillop) Date: Thu, 30 Apr 2009 02:46:44 -0700 (PDT) Subject: [Freeswitch-users] Ask for name in conferencing? Message-ID: <1241084804083-2746159.post@n2.nabble.com> As a former user of the app_confcall (http://www.freeswitch.org/node/100) Asterisk module produced by FreeSWITCH I'm in the process of moving a number of Asterisk-based services to FreeSWITCH and am trying to find the equivalent of the "record name before enter" feature of app_confcall. It seems strange that the FreeSWITCH conference module doesn't include this when the one built by the FreeSWITCH developers for Asterisk does... I know I could amend conference.js to do it, however there's no point in re-inventing the wheel if there's already another way. Anyone done this with FreeSWITCH, if so, how? Regards, Ross -- View this message in context: http://n2.nabble.com/Ask-for-name-in-conferencing--tp2746159p2746159.html Sent from the freeswitch-users mailing list archive at Nabble.com. From rossmck at mac.com Thu Apr 30 04:46:52 2009 From: rossmck at mac.com (Ross McKillop) Date: Thu, 30 Apr 2009 12:46:52 +0100 Subject: [Freeswitch-users] Ask for name in conferencing? Message-ID: <7D883B3A-FA3B-418D-98A6-41DAF5078E25@mac.com> As a former user of the app_confcall (http://www.freeswitch.org/node/ 100) Asterisk module produced by FreeSWITCH I'm in the process of moving a number of Asterisk-based services to FreeSWITCH and am trying to find the equivalent of the "record name before enter" feature of app_confcall. It seems strange that the FreeSWITCH conference module doesn't include this when the one built by the FreeSWITCH developers for Asterisk does... I know I could amend conference.js to do it, however there's no point in re-inventing the wheel if there's already another way. Anyone done this with FreeSWITCH, if so, how? Regards, Ross p.s. if it appears that i've posted this twice please accept my apologies - I tried through Nabble and it failed, so i've used a proper mail client this time... From daniel at rimspace.net Wed Apr 29 22:43:16 2009 From: daniel at rimspace.net (Daniel Pittman) Date: Thu, 30 Apr 2009 15:43:16 +1000 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> Message-ID: <87fxfqk87f.fsf@rimspace.net> Pawe? Pier?cionek writes: G'day Pawe?. > boot Your kernel with "divider=10 nohz=off" options :) > > Recent kernels are tickless which basically causes all freeswitch > timers/sleeps to fire at requested microsecond intervals. With nohz > kernels You get hundred times more system calls with freeswitch :( Like Jason, I am also interested to know why the tickless kernel causes the timers to generate so much more load. I can't find anything documented anywhere, really, about the issue ? and this thread is the only thing Google turns up on the topic. I am looking to move my SIP system to FreeSwitch some time soon, if I can, but I would love to know why nohz is so hostile to FreeSwitch before I do, if possible. Regards, Daniel From brian at freeswitch.org Thu Apr 30 06:18:18 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 30 Apr 2009 08:18:18 -0500 Subject: [Freeswitch-users] Ask for name in conferencing? In-Reply-To: <7D883B3A-FA3B-418D-98A6-41DAF5078E25@mac.com> References: <7D883B3A-FA3B-418D-98A6-41DAF5078E25@mac.com> Message-ID: <683E07D2-863B-4BF5-A60E-6F0BDA53DDED@freeswitch.org> Ross, I can see no reason to have the conference module do that for you when there are so many ways to do that externally with javascript, lua or any other the other languages then you can inject the sound file into the conference on demand before you drop the participant in. I like simplicity due to the fact it has less bugs /b On Apr 30, 2009, at 6:46 AM, Ross McKillop wrote: > As a former user of the app_confcall (http://www.freeswitch.org/node/ > 100) Asterisk module produced by FreeSWITCH I'm in the process of > moving a number of Asterisk-based services to FreeSWITCH and am trying > to find the equivalent of the "record name before enter" feature of > app_confcall. > > It seems strange that the FreeSWITCH conference module doesn't include > this when the one built by the FreeSWITCH developers for Asterisk > does... I know I could amend conference.js to do it, however there's > no point in re-inventing the wheel if there's already another way. > > Anyone done this with FreeSWITCH, if so, how? > > Regards, > Ross > > p.s. if it appears that i've posted this twice please accept my > apologies - I tried through Nabble and it failed, so i've used a > proper mail client this time... Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/a3af6239/attachment-0002.html From paul.degt at gmail.com Thu Apr 30 06:34:47 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 30 Apr 2009 09:34:47 -0400 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> Message-ID: <49F9A8F7.8050703@gmail.com> Worked for Grandstream, but not for X-Lite. Nik Middleton wrote: > Don't know where the setting is in FS, but force them to register every > 120 seconds and see if that helps > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 29 April 2009 20:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Phones become unreachable after some > time > > They do, but all necessary ports for FS are open. If that is fw issue, > are there ways to fight with it? > > Nik Middleton wrote: > >> Do the phones and FS have a firewall between them? If so, sounds like >> the pin hole in the fw is being closed. Alot only stay open for 4 >> > mins > >> Regards, >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> paul.degt >> Sent: 29 April 2009 20:15 >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Phones become unreachable after some time >> >> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds >> > > >> to Mysql DB for SIP registrations, presence etc. >> I noticed that after some time probably >30 min. phones which have >> > been > >> registered but without making calls become unreachable. Meaning that >> > any > >> call to such extension gets forwarded to VM as if it was offline, >> > until > >> I reload such phone. >> I did try to make the phones to register every 5 min. but it does not >> help. I also see valid registration information in sip_registrations >> table. X-Lite has r-port and keep alive settings on. >> Would appreciate any hints on what can be the issue here. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Richard.Lamkin at mettoni.com Thu Apr 30 07:24:21 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 30 Apr 2009 15:24:21 +0100 Subject: [Freeswitch-users] RFC compliance list, FS Road map and RFC.5411 Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804F44941@nickel.mettonigroup.com> Q1 - I have looked on the wiki and was unable to find a list of RFC's that FS is intended to comply with. The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a list of SIP protocols by name but these have no RFC number against them. Have I just missed the page?, if not is there any plan to put such a page together? Is there a SIP compliance matrix ? Q2 -This finally brings me on to my question; Are there any plans to publish an FS road map ? Even a wish list of features which users of FS vote on would be helpful. I know FS is OSS and it does fall to all and not just the core team to implement/extend the product but a road map would Q3- I have recently been looking at RFC.5411 which is a basically list of SIP RFC's. A developer or system designer like me is in the future are likely to use an RFC like it as a compliance list for SIP stack selection. Ultimately, it will be the likes of marketing men who will be looking for the one stop shop for a SIP spec who use RFC5411 as the SIP part of a product spec. Are there any plans to use RFC 5411 as a goal? Best Regards Richard Lamkin Mettoni Group UK richard.lamkin at mettonigroup.com ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/d6ed3788/attachment-0002.html From paul.degt at gmail.com Thu Apr 30 07:45:21 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 30 Apr 2009 10:45:21 -0400 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> Message-ID: <49F9B981.707@gmail.com> Correction: 2 min. registration timeout does not work for either Grandstream 386 nor for X-Lite. Will try 1 min., but I am skeptical. Grandstream has other weird issues btw, like not getting dial tone from first attempt or sometimes giving buzzing noise instead of one. My fw is fairly old Netgear unit, would newer models be better in this area, or I need SIP-aware one? Nik Middleton wrote: > Don't know where the setting is in FS, but force them to register every > 120 seconds and see if that helps > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 29 April 2009 20:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Phones become unreachable after some > time > > They do, but all necessary ports for FS are open. If that is fw issue, > are there ways to fight with it? > > Nik Middleton wrote: > >> Do the phones and FS have a firewall between them? If so, sounds like >> the pin hole in the fw is being closed. Alot only stay open for 4 >> > mins > >> Regards, >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> paul.degt >> Sent: 29 April 2009 20:15 >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Phones become unreachable after some time >> >> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds >> > > >> to Mysql DB for SIP registrations, presence etc. >> I noticed that after some time probably >30 min. phones which have >> > been > >> registered but without making calls become unreachable. Meaning that >> > any > >> call to such extension gets forwarded to VM as if it was offline, >> > until > >> I reload such phone. >> I did try to make the phones to register every 5 min. but it does not >> help. I also see valid registration information in sip_registrations >> table. X-Lite has r-port and keep alive settings on. >> Would appreciate any hints on what can be the issue here. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 30 07:46:31 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 30 Apr 2009 07:46:31 -0700 (PDT) Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <1241015506.11362.1.camel@portable-evil> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> Message-ID: <23317579.post@talk.nabble.com> Thanks guys for the links on CF-to-IDE adaptors. -- View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23317579.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gallo at mctelefonia.com Thu Apr 30 07:51:34 2009 From: gallo at mctelefonia.com (Antonio Gallo) Date: Thu, 30 Apr 2009 16:51:34 +0200 Subject: [Freeswitch-users] RFC compliance list, FS Road map and RFC.5411 In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804F44941@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804F44941@nickel.mettonigroup.com> Message-ID: <49F9BAF6.7080001@mctelefonia.com> Richard Lamkin ha scritto: > > Q1 -- I have looked on the wiki and was unable to find a list of RFC's > that FS is intended to comply with. > > The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a > list of SIP protocols by name but these have no RFC number against them. > > Have I just missed the page?, if not is there any plan to put such a > page together? > > Is there a SIP compliance matrix ? > AFAIK it uses Sofia SIP library and this is the link to the library itself implemented stuffs http://sofia-sip.sourceforge.net/refdocs/sofia_sip_conformance.html You get all the RFC numbers you want here :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/8f16f5cc/attachment-0002.html From rossmck at mac.com Thu Apr 30 08:20:51 2009 From: rossmck at mac.com (Ross McKillop) Date: Thu, 30 Apr 2009 08:20:51 -0700 (PDT) Subject: [Freeswitch-users] Ask for name in conferencing? In-Reply-To: <683E07D2-863B-4BF5-A60E-6F0BDA53DDED@freeswitch.org> References: <7D883B3A-FA3B-418D-98A6-41DAF5078E25@mac.com> <683E07D2-863B-4BF5-A60E-6F0BDA53DDED@freeswitch.org> Message-ID: <1241104851593-2747788.post@n2.nabble.com> Brian West wrote: > > Ross, > I can see no reason to have the conference module do that for you > when there are so many ways to do that externally with javascript, lua > or any other the other languages then you can inject the sound file > into the conference on demand before you drop the participant in. I > like simplicity due to the fact it has less bugs > > /b > Thanks for the quick response ... will do it with JS... I've just not managed to get my head around Lua yet ;) Whilst it's not an ideal language it'd be quite nice to be able to use PHP with FreeSWITCH ... I've got literally hundreds of AGIs to convert and it'd make it a lot easier ... but that's for another day and another thread. Thanks again, Ross -- View this message in context: http://n2.nabble.com/Ask-for-name-in-conferencing--tp2746986p2747788.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Thu Apr 30 08:43:27 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 30 Apr 2009 11:43:27 -0400 Subject: [Freeswitch-users] RFC compliance list, FS Road map and RFC.5411 In-Reply-To: <49F9BAF6.7080001@mctelefonia.com> References: <3181A30B8C35AB4AA8577B78DDF4613804F44941@nickel.mettonigroup.com> <49F9BAF6.7080001@mctelefonia.com> Message-ID: I can confirm that we comply with rfc 5411 in that we agree that is a list of sip specs that we may or may not honor, and that we may or may not have ever seen or read. Joking aside, the sofia list is pretty good, there are some things noted as it would be implemented in the application. If you have specific questions about things notated like that in the sofia link below, reply to this thread and we'll try to sort out if we intend or possibly do support it. Mike On Apr 30, 2009, at 10:51 AM, Antonio Gallo wrote: > Richard Lamkin ha scritto: >> >> Q1 ? I have looked on the wiki and was unable to find a list of >> RFC?s that FS is intended to comply with. >> The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a >> list of SIP protocols by name but these have no RFC number against >> them. >> Have I just missed the page?, if not is there any plan to put such >> a page together? >> Is there a SIP compliance matrix ? > AFAIK it uses Sofia SIP library and this is the link to the library > itself implemented stuffs > http://sofia-sip.sourceforge.net/refdocs/sofia_sip_conformance.html > > You get all the RFC numbers you want here :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/9b3531ce/attachment-0002.html From Richard.Lamkin at mettoni.com Thu Apr 30 08:47:35 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 30 Apr 2009 16:47:35 +0100 Subject: [Freeswitch-users] RFC compliance list, FS Road map and RFC.5411 In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380472F995@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF461380472F995@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804F44A1C@nickel.mettonigroup.com> Sorry in my eagerness I seem to have sent my original email with a few words missing. Too much cut and paste! Q1 - I have looked on the wiki and was unable to find a list of RFC's that FS is intended to comply with. The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a list of SIP protocols by name but these have no RFC number against them. Have I just missed the page?, if not is there any plan to put such a page together? Is there a SIP compliance matrix ? Q2 -Are there any plans to publish an FS road map ? Even a wish list of features which users of FS vote on would be helpful. I know FS is OSS and it does fall to all and not just the core team to implement/extend the product but a road map would be helpful. Q3- I have recently been looking at RFC.5411 which is a basically list of SIP RFC's. A developer or system designer like me is in the future are likely to use an RFC like it as a compliance list for SIP stack selection. Ultimately, it will be the likes of marketing men who will be looking for the one stop shop for a SIP spec who use RFC5411 as the SIP part of a product spec. Are there any plans to use RFC 5411 as a goal? Best Regards Richard Lamkin Mettoni Group UK richard.lamkin at mettonigroup.com ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/29d1871a/attachment-0002.html From anthony.minessale at gmail.com Thu Apr 30 08:55:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 30 Apr 2009 10:55:59 -0500 Subject: [Freeswitch-users] Ask for name in conferencing? In-Reply-To: <1241104851593-2747788.post@n2.nabble.com> References: <7D883B3A-FA3B-418D-98A6-41DAF5078E25@mac.com> <683E07D2-863B-4BF5-A60E-6F0BDA53DDED@freeswitch.org> <1241104851593-2747788.post@n2.nabble.com> Message-ID: <191c3a030904300855t11856208yc02328ca0af33b96@mail.gmail.com> PHP works with ESL which not entirely unlike AGI On Thu, Apr 30, 2009 at 10:20 AM, Ross McKillop wrote: > > > Brian West wrote: > > > > Ross, > > I can see no reason to have the conference module do that for you > > when there are so many ways to do that externally with javascript, lua > > or any other the other languages then you can inject the sound file > > into the conference on demand before you drop the participant in. I > > like simplicity due to the fact it has less bugs > > > > /b > > > > Thanks for the quick response ... will do it with JS... I've just not > managed to get my head around Lua yet ;) > > Whilst it's not an ideal language it'd be quite nice to be able to use PHP > with FreeSWITCH ... I've got literally hundreds of AGIs to convert and it'd > make it a lot easier ... but that's for another day and another thread. > > Thanks again, > Ross > > -- > View this message in context: > http://n2.nabble.com/Ask-for-name-in-conferencing--tp2746986p2747788.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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/4ab23277/attachment-0002.html From pawel at voiceworks.pl Thu Apr 30 11:28:23 2009 From: pawel at voiceworks.pl (=?UTF-8?Q?Pawe=C5=82_Pier=C5=9Bcionek?=) Date: Thu, 30 Apr 2009 20:28:23 +0200 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel In-Reply-To: <87fxfqk87f.fsf@rimspace.net> References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> <87fxfqk87f.fsf@rimspace.net> Message-ID: <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> Hi, With really old kernels (100Hz) if You do sleep(1ms) You sleep for 10ms on average. With enterprise kernels (250Hz) Your sleep resolution increases by a factor of 4. With fresh kernels (1000Hz) You get real 1ms timer resolution - 10fold increase compared to old kernels. With tickless You get whatever resolution You want - eg when You sleep for 100 microseconds(micro not mili) then You get exactly what You wish for. Now for reasons I do no try to understand :) there are a lot of really short sleeps and fast timers in FreeSwitch - like 100 micro(1/10th of a ms). So with CentOS such a 100 microsecond sleep cannot "fire" faster then 250 times a second. With tickless kernel same 100 microsecond sleep "fires" 10k times a second. Pawel, From brian at freeswitch.org Thu Apr 30 11:36:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 30 Apr 2009 13:36:56 -0500 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel In-Reply-To: <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> <87fxfqk87f.fsf@rimspace.net> <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> Message-ID: Then it would be recommended to not do tickless clock :P /b On Apr 30, 2009, at 1:28 PM, Pawe? Pier?cionek wrote: > Hi, > > With really old kernels (100Hz) if You do sleep(1ms) You sleep for > 10ms on average. > With enterprise kernels (250Hz) Your sleep resolution increases by a > factor of 4. > With fresh kernels (1000Hz) You get real 1ms timer resolution - > 10fold increase compared to old kernels. > > With tickless You get whatever resolution You want - eg when You > sleep for 100 microseconds(micro not mili) then You get exactly what > You wish for. > > Now for reasons I do no try to understand :) there are a lot of > really short sleeps and fast timers in FreeSwitch - like 100 > micro(1/10th of a ms). > So with CentOS such a 100 microsecond sleep cannot "fire" faster > then 250 times a second. > With tickless kernel same 100 microsecond sleep "fires" 10k times a > second. > > Pawel, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/9b0eacd4/attachment-0002.html From mike at jerris.com Thu Apr 30 11:48:21 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 30 Apr 2009 14:48:21 -0400 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel In-Reply-To: <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> <87fxfqk87f.fsf@rimspace.net> <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> Message-ID: <636F7D02-E6E2-419F-9F96-AB2AC1A893F8@jerris.com> Can you point out any place we do sub milli second sleeps? The timer thread should be doing 1ms, I can't think of any that would be less. MIke On Apr 30, 2009, at 2:28 PM, Pawe? Pier?cionek wrote: > Hi, > > With really old kernels (100Hz) if You do sleep(1ms) You sleep for > 10ms on average. > With enterprise kernels (250Hz) Your sleep resolution increases by a > factor of 4. > With fresh kernels (1000Hz) You get real 1ms timer resolution - > 10fold increase compared to old kernels. > > With tickless You get whatever resolution You want - eg when You > sleep for 100 microseconds(micro not mili) then You get exactly what > You wish for. > > Now for reasons I do no try to understand :) there are a lot of > really short sleeps and fast timers in FreeSwitch - like 100 > micro(1/10th of a ms). > So with CentOS such a 100 microsecond sleep cannot "fire" faster > then 250 times a second. > With tickless kernel same 100 microsecond sleep "fires" 10k times a > second. > > Pawel, From nik.middleton at noblesolutions.co.uk Thu Apr 30 11:54:42 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 30 Apr 2009 19:54:42 +0100 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: <49F9A8F7.8050703@gmail.com> References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> Message-ID: Xlite may be working on the timeout FS is sending. See the following from the wiki and see if that helps, but I'm not sure In domain, set -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of paul.degt Sent: 30 April 2009 14:35 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Phones become unreachable after some time Worked for Grandstream, but not for X-Lite. Nik Middleton wrote: > Don't know where the setting is in FS, but force them to register every > 120 seconds and see if that helps > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 29 April 2009 20:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Phones become unreachable after some > time > > They do, but all necessary ports for FS are open. If that is fw issue, > are there ways to fight with it? > > Nik Middleton wrote: > >> Do the phones and FS have a firewall between them? If so, sounds like >> the pin hole in the fw is being closed. Alot only stay open for 4 >> > mins > >> Regards, >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> paul.degt >> Sent: 29 April 2009 20:15 >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Phones become unreachable after some time >> >> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds >> > > >> to Mysql DB for SIP registrations, presence etc. >> I noticed that after some time probably >30 min. phones which have >> > been > >> registered but without making calls become unreachable. Meaning that >> > any > >> call to such extension gets forwarded to VM as if it was offline, >> > until > >> I reload such phone. >> I did try to make the phones to register every 5 min. but it does not >> help. I also see valid registration information in sip_registrations >> table. X-Lite has r-port and keep alive settings on. >> Would appreciate any hints on what can be the issue here. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 gallo at mctelefonia.com Thu Apr 30 12:47:57 2009 From: gallo at mctelefonia.com (Antonio Gallo - MC) Date: Thu, 30 Apr 2009 21:47:57 +0200 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> Message-ID: <49FA006D.8080200@mctelefonia.com> Nik Middleton ha scritto: > Xlite may be working on the timeout FS is sending. > If Xlite is monitoring any user when this user reboot/unregister (then Xlite get a publish_out event) when that user come back online then Xlite is never notified until it re-register/or re-start Anyway i don't care much about Xlite i just used it for testing from home with the office machine. From chris at fowler.cc Thu Apr 30 13:54:01 2009 From: chris at fowler.cc (Chris Fowler) Date: Thu, 30 Apr 2009 16:54:01 -0400 Subject: [Freeswitch-users] Audio delay when conferencing In-Reply-To: <49FA006D.8080200@mctelefonia.com> References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> <49FA006D.8080200@mctelefonia.com> Message-ID: <7454A296C7EDE34EA57199FAA401E2F114DB7AE069@VMBX113.ihostexchange.net> I'm using FreeSWITCH (Build 13168M) and we're having intermittent multi-second delays on conference bridges with more than three participants (this is not a new issue - just bubbled to the top of the stack to address). The server is running on Amazon's AWS c1.medium instance, CentOS 5.0 with Kernel 2.6.18 32-bit i386. I recorded a conference which shows the problem nicely: http://cfowl.postinbox.com/c.wav Callers are coming on via the internal sofia profile from various physical locations. I'm not sure how to proceed with debugging this issue - advice welcome. Thanks, Chris. From paul.degt at gmail.com Thu Apr 30 14:28:35 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 30 Apr 2009 17:28:35 -0400 Subject: [Freeswitch-users] Phones become unreachable after some time In-Reply-To: References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> Message-ID: <49FA1803.8050601@gmail.com> Somehow 60 sec. interval works on both phones just fine. Appreciate everybody's input. Nik Middleton wrote: > Xlite may be working on the timeout FS is sending. > > See the following from the wiki and see if that helps, but I'm not sure > > > In domain, set > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > paul.degt > Sent: 30 April 2009 14:35 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Phones become unreachable after some > time > > Worked for Grandstream, but not for X-Lite. > > Nik Middleton wrote: > >> Don't know where the setting is in FS, but force them to register >> > every > >> 120 seconds and see if that helps >> >> Regards, >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> paul.degt >> Sent: 29 April 2009 20:50 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Phones become unreachable after some >> time >> >> They do, but all necessary ports for FS are open. If that is fw issue, >> > > >> are there ways to fight with it? >> >> Nik Middleton wrote: >> >> >>> Do the phones and FS have a firewall between them? If so, sounds >>> > like > >>> the pin hole in the fw is being closed. Alot only stay open for 4 >>> >>> >> mins >> >> >>> Regards, >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>> paul.degt >>> Sent: 29 April 2009 20:15 >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: [Freeswitch-users] Phones become unreachable after some time >>> >>> I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS >>> > binds > >>> >>> >> >> >>> to Mysql DB for SIP registrations, presence etc. >>> I noticed that after some time probably >30 min. phones which have >>> >>> >> been >> >> >>> registered but without making calls become unreachable. Meaning that >>> >>> >> any >> >> >>> call to such extension gets forwarded to VM as if it was offline, >>> >>> >> until >> >> >>> I reload such phone. >>> I did try to make the phones to register every 5 min. but it does not >>> > > >>> help. I also see valid registration information in sip_registrations >>> table. X-Lite has r-port and keep alive settings on. >>> Would appreciate any hints on what can be the issue here. >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >>> http://www.freeswitch.org >>> >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Apr 30 14:30:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 30 Apr 2009 16:30:43 -0500 Subject: [Freeswitch-users] Audio delay when conferencing In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F114DB7AE069@VMBX113.ihostexchange.net> References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> <49FA006D.8080200@mctelefonia.com> <7454A296C7EDE34EA57199FAA401E2F114DB7AE069@VMBX113.ihostexchange.net> Message-ID: <191c3a030904301430q579b44fex3165e10bc5b71a9d@mail.gmail.com> the mailing list is not the correct place to report issues. http://jira.freeswitch.org We have 8 people conferences that last as long as 12 hours a day every day and there is no delay. Be advised if you open a jira it will require that you download and compile and retest your issue on SVN trunk. If we cannot find an issue in FreeSWITCH, consider contacting FreeSWITCH Solutions for commercial support debugging your 3rd party elements: http://www.freeswitchsolutions.com On Thu, Apr 30, 2009 at 3:54 PM, Chris Fowler wrote: > I'm using FreeSWITCH (Build 13168M) and we're having intermittent > multi-second delays on conference bridges with more than three participants > (this is not a new issue - just bubbled to the top of the stack to address). > > The server is running on Amazon's AWS c1.medium instance, CentOS 5.0 with > Kernel 2.6.18 32-bit i386. > > I recorded a conference which shows the problem nicely: > http://cfowl.postinbox.com/c.wav > > Callers are coming on via the internal sofia profile from various physical > locations. I'm not sure how to proceed with debugging this issue - advice > welcome. > > > > > > > > > > > > > > > > value="tone_stream://%(500,0,300,200,100,50,25)"/> > > > > value="conference/conf-is-unlocked.wav"/> > > > > > > > > > Thanks, Chris. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/f2008b39/attachment-0002.html From brian at freeswitch.org Thu Apr 30 14:32:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 30 Apr 2009 16:32:51 -0500 Subject: [Freeswitch-users] Audio delay when conferencing In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F114DB7AE069@VMBX113.ihostexchange.net> References: <49F8A750.7030906@gmail.com> <49F8AF74.5040004@gmail.com> <49F9A8F7.8050703@gmail.com> <49FA006D.8080200@mctelefonia.com> <7454A296C7EDE34EA57199FAA401E2F114DB7AE069@VMBX113.ihostexchange.net> Message-ID: <7585F4EF-E4FD-4E71-BCDC-B76455B9FD33@freeswitch.org> Have you tried on non-ec2 installs? Maybe some setting on the EC2 instance is messing with it. Also don't hijack threads please! :) /b On Apr 30, 2009, at 3:54 PM, Chris Fowler wrote: > I'm using FreeSWITCH (Build 13168M) and we're having intermittent > multi-second delays on conference bridges with more than three > participants (this is not a new issue - just bubbled to the top of > the stack to address). > > The server is running on Amazon's AWS c1.medium instance, CentOS 5.0 > with Kernel 2.6.18 32-bit i386. > > I recorded a conference which shows the problem nicely: http://cfowl.postinbox.com/c.wav > > Callers are coming on via the internal sofia profile from various > physical locations. I'm not sure how to proceed with debugging this > issue - advice welcome. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/549ae6d8/attachment-0002.html From can_man at gmx.de Thu Apr 30 15:37:01 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Fri, 01 May 2009 00:37:01 +0200 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error Message-ID: <20090430223701.280500@gmx.net> Hello, I am trying to get skypiax working, but I am having trouble with the sound. The calls fail with CALL FAILUREREASON 7 = Sound I/O error and I am getting the following error: ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi I am running centos 5.3 and have followed the installation guide on the wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When saving the configuration on my desktop I have set the sound card to snd_dummy. On the server the startup script load snd-dumy like this /sbin/modprobe snd-dummy enable=1. Below is the output of lsmod and the debug output from FS. It would be great if someone could help me fix my problem. Thank you very much. Best wishes, Phil -bash-3.2# lsmod Module Size Used by snd_dummy 12416 0 snd_seq_oss 32832 0 snd_seq_midi_event 7744 1 snd_seq_oss snd_seq 55200 4 snd_seq_oss,snd_seq_midi_event snd_seq_device 7120 1 snd_seq_oss snd_pcm_oss 44480 0 snd_mixer_oss 16512 1 snd_pcm_oss snd_pcm 79624 2 snd_dummy,snd_pcm_oss snd_timer 22088 2 snd_seq,snd_pcm snd 55976 8 snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 7456 1 snd snd_page_alloc 8720 1 snd_pcm freeswitch at voipserverServerFreeswitch> load mod_skypiax 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 718 ][none ][-1,-1,-1] globals.debug=0 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 720 ][none ][-1,-1,-1] globals.debug=8 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 731 ][none ][-1,-1,-1] codec-master globals.debug=8 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 734 ][none ][-1,-1,-1] globals.dialplan=XML 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 740 ][none ][-1,-1,-1] globals.context=default 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 743 ][none ][-1,-1,-1] globals.codec_string=gsm,ulaw 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 750 ][none ][-1,-1,-1] globals.codec_rates_string=8000,16000 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 723 ][none ][-1,-1,-1] globals.hold_music= 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 737 ][none ][-1,-1,-1] globals.destination=5000 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] interface_id=1 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] name=skypiax1 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized XInitThreads! 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax1 ][-1, 0, 0] CONFIGURING interface_id=1 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].destination=3101 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax1 ][-1, 0, 0] interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev 13177[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING interface_id=1 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 1407 ][skypiax1 ][-1, 0, 0] X Display ':101' opened 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype instance found with id #2097454 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 ][skypiax1 ][-1, 0, 0] In skypiax_signaling_thread_func: started, p=0x2aaab93226f8 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||OK||| 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||PROTOCOL 7||| 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||CURRENTUSERHANDLE xyzUK||| 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax1 ][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: xyzUK, skype_user: xyzUK! 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||USERSTATUS ONLINE||| 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev 13177[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev 13177[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=1, waiting 60 seconds for CURRENTUSERHANDLE==xyzUK 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev 13177[(nil)|37 ][WARNINGA 1004 ][skypiax1 ][-1, 0, 0] Interface_id=1 is now STARTED, the Skype client to which we are connected gave us the correct CURRENTUSERHANDLE (xyzUK) 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] interface_id=2 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] name=skypiax2 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized XInitThreads! 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax2 ][-1, 0, 0] CONFIGURING interface_id=2 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].destination=5000 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax2 ][-1, 0, 0] interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev 13177[(nil)|37 ][WARNINGA 950 ][skypiax2 ][-1, 0, 0] STARTING interface_id=2 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 1407 ][skypiax2 ][-1, 0, 0] X Display ':102' opened 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype instance found with id #2097454 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 ][skypiax2 ][-1, 0, 0] In skypiax_signaling_thread_func: started, p=0x2aaab9325c18 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: |||OK||| 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: |||PROTOCOL 7||| 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: |||CURRENTUSERHANDLE voipserver||| 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax2 ][-1, 0, 0] Skype MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: voipserver, skype_user: voipserver! 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: |||USERSTATUS ONLINE||| 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev 13177[(nil)|37 ][NOTICA 976 ][skypiax2 ][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=2 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev 13177[(nil)|37 ][NOTICA 986 ][skypiax2 ][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=2, waiting 60 seconds for CURRENTUSERHANDLE==voipserver API CALL [load(mod_skypiax)] output: +OK 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev 13177[(nil)|37 ][WARNINGA 1004 ][skypiax2 ][-1, 0, 0] Interface_id=2 is now STARTED, the Skype client to which we are connected gave us the correct CURRENTUSERHANDLE (voipserver) 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].interface_id=1 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].X11_display=:101 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].name=skypiax1 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].context=default 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].dialplan=XML 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].destination=3101 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax1 ][-1, 0, 0] i=1 globals.SKYPIAX_INTERFACES[1].context=default 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].interface_id=2 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].X11_display=:102 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].name=skypiax2 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].context=default 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].dialplan=XML 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].destination=5000 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax2 ][-1, 0, 0] i=2 globals.SKYPIAX_INTERFACES[2].context=default 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_skypiax] 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 switch_loadable_module_process() Adding Endpoint 'skypiax' 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'sk' 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'skypiax' freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||USER paolofun6 PHONE_MOBILE +420775216536||| freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/external/07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel sofia/external/07771236762 at sipgate.co.uk entering state [received][100] 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 15141 15141 IN IP4 217.10.66.71 s=session c=IN IP4 217.10.66.71 t=0 0 m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 sofia_glue_tech_set_codec() Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms 160 samples 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 sofia_handle_sip_i_state() (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> CS_INIT 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal sofia/external/07771236762 at sipgate.co.uk [BREAK] 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) Running State Change CS_INIT 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State INIT 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/external/07771236762 at sipgate.co.uk SOFIA INIT 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> CS_ROUTING 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal sofia/external/07771236762 at sipgate.co.uk [BREAK] 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State INIT going to sleep 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) Running State Change CS_ROUTING 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State ROUTING 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 switch_core_standard_on_routing() sofia/external/07771236762 at sipgate.co.uk Standard ROUTING 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 07771236762->00442083324655 in context public Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing [public->skype_uri] continue=false Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) [skype_uri] destination_number(00442083324655) =~ /^(00442083324655)$/ break=on-false Dialplan: sofia/external/07771236762 at sipgate.co.uk Action bridge(skypiax/skypiax1/xyzTestUK) 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/external/07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal sofia/external/07771236762 at sipgate.co.uk [BREAK] 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State ROUTING going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) Running State Change CS_EXECUTE 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State EXECUTE 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/external/07771236762 at sipgate.co.uk Standard EXECUTE EXECUTE sofia/external/07771236762 at sipgate.co.uk bridge(skypiax/skypiax1/xyzTestUK) 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 channel_outgoing_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 585 ][ ][-1, 0, 0] globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() rev 13177[(nil)|37 ][DEBUG_SKYPE 151 ][skypiax1 ][-1, 0, 0] skypiax_codec SUCCESS 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/skypiax1/xyzTestUK [0375c668-b4a2-4364-a8c6-0a718d4f00a3] 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() rev 13177[(nil)|37 ][DEBUG_SKYPE 773 ][skypiax1 ][-1, 0, 0] Calling Skype, rdest is: xyzTestUK 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 ][skypiax1 ][-1, 0, 0] SENDING: |||SET AGC OFF|||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 ][skypiax1 ][-1, 0, 0] SENDING: |||SET AEC OFF|||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 ][skypiax1 ][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 channel_outgoing_channel() (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal skypiax/skypiax1/xyzTestUK [BREAK] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change CS_INIT 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal skypiax/skypiax1/xyzTestUK [BREAK] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() rev 13177[(nil)|37 ][DEBUG_SKYPE 182 ][skypiax1 ][-1, 0, 0] skypiax/skypiax1/xyzTestUK CHANNEL INIT 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change CS_ROUTING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() rev 13177[(nil)|37 ][DEBUG_SKYPE 257 ][skypiax1 ][-1, 0, 0] skypiax/skypiax1/xyzTestUK CHANNEL ROUTING 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal skypiax/skypiax1/xyzTestUK [BREAK] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change CS_CONSUME_MEDIA 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||AGC OFF||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||AEC OFF||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||CALL 455 STATUS UNPLACED||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 0, 0] Skype MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: UNPLACED,where: NULL! ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 371 ][skypiax1 ][-1, 3,116] skype_call: 455 is now UNPLACED ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,116] READING: |||CALL 455 STATUS ROUTING||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,116] Skype MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: ROUTING,where: NULL! 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 365 ][skypiax1 ][-1, 3,117] skype_call: 455 is now ROUTING 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: |||CALL 455 FAILUREREASON 7||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, value: 7,where: NULL! 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 201 ][skypiax1 ][-1, 3,117] Skype FAILED on skype_call 455. Let's wait for the FAILED message. 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: |||CALL 455 VAA_INPUT_STATUS FALSE||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, value: FALSE,where: NULL! 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: |||CALL 455 STATUS FAILED||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: FAILED,where: NULL! 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 334 ][skypiax1 ][-1, 3,112] we tried to call Skype on skype_call 455 and Skype has now FAILED 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 ][skypiax1 ][-1, 1,112] skype call ended 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK [CS_CONSUME_MEDIA] [NORMAL_CLEARING] 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 switch_channel_perform_hangup() Send signal skypiax/skypiax1/xyzTestUK [KILL] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 293 ][skypiax1 ][-1, 1,112] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal skypiax/skypiax1/xyzTestUK [BREAK] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 switch_ivr_originate() Originate Resulted in Error Cause: 16 [NORMAL_CLEARING] 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 audio_bridge_function() Originate Failed. Cause: NORMAL_CLEARING 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 audio_bridge_function() Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 switch_channel_perform_hangup() Send signal sofia/external/07771236762 at sipgate.co.uk [KILL] 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal sofia/external/07771236762 at sipgate.co.uk [BREAK] 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State EXECUTE going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) Running State Change CS_HANGUP 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State HANGUP 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/external/07771236762 at sipgate.co.uk hanging up, cause: NORMAL_CLEARING 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 480 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/07771236762 at sipgate.co.uk Standard HANGUP, cause: NORMAL_CLEARING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State HANGUP going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State Change CS_HANGUP -> CS_REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal sofia/external/07771236762 at sipgate.co.uk [BREAK] 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) Running State Change CS_REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 switch_core_session_reporting_state() (sofia/external/07771236762 at sipgate.co.uk) State REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change CS_HANGUP 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev 13177[(nil)|37 ][DEBUG_SKYPE 228 ][skypiax1 ][-1, 1,112] hanging up skype call: 455 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 ][skypiax1 ][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev 13177[(nil)|37 ][DEBUG_SKYPE 235 ][skypiax1 ][-1, 1,112] skypiax/skypiax1/xyzTestUK CHANNEL HANGUP 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard HANGUP, cause: NORMAL_CLEARING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change CS_HANGUP -> CS_REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 switch_core_session_signal_state_change() Send signal skypiax/skypiax1/xyzTestUK [BREAK] 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change CS_REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State REPORTING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard REPORTING, cause: NORMAL_CLEARING 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State REPORTING going to sleep 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change CS_REPORTING -> CS_DESTROY 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Locked, Waiting on external entities 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Ended 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK [CS_DESTROY] 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State DESTROY 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard DESTROY 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State DESTROY going to sleep 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 1,112] READING: |||ERROR 559 CALL: Action failed||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 91 ][skypiax1 ][-1, 1,112] Skype got ERROR: |||ERROR||| 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 skypiax_signaling_read() rev 13177[(nil)|37 ][DEBUG_SKYPE 93 ][skypiax1 ][-1, 1,110] skype_call now is DOWN 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 ][skypiax1 ][-1, 1,110] skype call ended 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 687 ][skypiax1 ][-1, 1,110] no session 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/external/07771236762 at sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 switch_core_session_reporting_state() (sofia/external/07771236762 at sipgate.co.uk) State REPORTING going to sleep 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State Change CS_REPORTING -> CS_DESTROY 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 switch_core_session_thread() Session 1 (sofia/external/07771236762 at sipgate.co.uk) Locked, Waiting on external entities 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 switch_core_session_thread() Session 1 (sofia/external/07771236762 at sipgate.co.uk) Ended 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 switch_core_session_thread() Close Channel sofia/external/07771236762 at sipgate.co.uk [CS_DESTROY] 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/external/07771236762 at sipgate.co.uk) State DESTROY 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/external/07771236762 at sipgate.co.uk Standard DESTROY 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/external/07771236762 at sipgate.co.uk) State DESTROY going to sleep -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a From anthony.minessale at gmail.com Thu Apr 30 16:02:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 30 Apr 2009 18:02:03 -0500 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error In-Reply-To: <20090430223701.280500@gmx.net> References: <20090430223701.280500@gmx.net> Message-ID: <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> if you put that info in a jira ticket http://jira.freeswitch.org and route it to skypeiax , the guy who maintains that module will see it. On Thu, Apr 30, 2009 at 5:37 PM, wrote: > > Hello, > > I am trying to get skypiax working, but I am having trouble with the sound. > The calls fail with CALL FAILUREREASON 7 = Sound I/O error and > I am getting the following error: > > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > > > I am running centos 5.3 and have followed the installation guide on the > wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When > saving > the configuration on my desktop I have set the sound card to snd_dummy. On > the server the startup script load snd-dumy like this /sbin/modprobe > snd-dummy enable=1. > Below is the output of lsmod and the debug output from FS. It would be > great if someone could help me fix my problem. > > Thank you very much. > Best wishes, > Phil > > > > > -bash-3.2# lsmod > Module Size Used by > snd_dummy 12416 0 > snd_seq_oss 32832 0 > snd_seq_midi_event 7744 1 snd_seq_oss > snd_seq 55200 4 snd_seq_oss,snd_seq_midi_event > snd_seq_device 7120 1 snd_seq_oss > snd_pcm_oss 44480 0 > snd_mixer_oss 16512 1 snd_pcm_oss > snd_pcm 79624 2 snd_dummy,snd_pcm_oss > snd_timer 22088 2 snd_seq,snd_pcm > snd 55976 8 > snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer > soundcore 7456 1 snd > snd_page_alloc 8720 1 snd_pcm > > > > freeswitch at voipserverServerFreeswitch> load mod_skypiax > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 718 ][none ][-1,-1,-1] > globals.debug=0 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 720 ][none ][-1,-1,-1] > globals.debug=8 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 731 ][none ][-1,-1,-1] codec-master > globals.debug=8 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 734 ][none ][-1,-1,-1] > globals.dialplan=XML > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 740 ][none ][-1,-1,-1] > globals.context=default > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 743 ][none ][-1,-1,-1] > globals.codec_string=gsm,ulaw > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 750 ][none ][-1,-1,-1] > globals.codec_rates_string=8000,16000 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 723 ][none ][-1,-1,-1] > globals.hold_music= > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 737 ][none ][-1,-1,-1] > globals.destination=5000 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] > interface_id=1 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] name=skypiax1 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized > XInitThreads! > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax1 ][-1, 0, 0] CONFIGURING > interface_id=1 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].destination=3101 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax1 ][-1, 0, 0] > interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default > 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev > 13177[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING > interface_id=1 > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 > skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 1407 > ][skypiax1 ][-1, 0, 0] X Display ':101' opened > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype > instance found with id #2097454 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 > skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 > ][skypiax1 ][-1, 0, 0] In skypiax_signaling_thread_func: started, > p=0x2aaab93226f8 > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||OK||| > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||PROTOCOL 7||| > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||CONNSTATUS ONLINE||| > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||CURRENTUSERHANDLE xyzUK||| > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax1 ][-1, 0, 0] Skype > MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: > xyzUK, skype_user: xyzUK! > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||USERSTATUS ONLINE||| > 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev > 13177[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 > seconds to find a running Skype client and connect to its SKYPE API for > interface_id=1 > 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev > 13177[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running > Skype client, connected to its SKYPE API for interface_id=1, waiting 60 > seconds for CURRENTUSERHANDLE==xyzUK > 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev > 13177[(nil)|37 ][WARNINGA 1004 ][skypiax1 ][-1, 0, 0] Interface_id=1 > is now STARTED, the Skype client to which we are connected gave us the > correct CURRENTUSERHANDLE (xyzUK) > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] > interface_id=2 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] name=skypiax2 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized > XInitThreads! > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax2 ][-1, 0, 0] CONFIGURING > interface_id=2 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax2 ][-1, 0, 0] > interface_id=2 > globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].destination=5000 > 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax2 ][-1, 0, 0] > interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default > 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev > 13177[(nil)|37 ][WARNINGA 950 ][skypiax2 ][-1, 0, 0] STARTING > interface_id=2 > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 > skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 1407 > ][skypiax2 ][-1, 0, 0] X Display ':102' opened > 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype > instance found with id #2097454 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 > skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 > ][skypiax2 ][-1, 0, 0] In skypiax_signaling_thread_func: started, > p=0x2aaab9325c18 > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: > |||OK||| > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: > |||PROTOCOL 7||| > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: > |||CONNSTATUS ONLINE||| > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: > |||CURRENTUSERHANDLE voipserver||| > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax2 ][-1, 0, 0] Skype > MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, cuh: > voipserver, skype_user: voipserver! > 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] READING: > |||USERSTATUS ONLINE||| > 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev > 13177[(nil)|37 ][NOTICA 976 ][skypiax2 ][-1, 0, 0] WAITING roughly 10 > seconds to find a running Skype client and connect to its SKYPE API for > interface_id=2 > 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev > 13177[(nil)|37 ][NOTICA 986 ][skypiax2 ][-1, 0, 0] Found a running > Skype client, connected to its SKYPE API for interface_id=2, waiting 60 > seconds for CURRENTUSERHANDLE==voipserver > API CALL [load(mod_skypiax)] output: > +OK > > 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev > 13177[(nil)|37 ][WARNINGA 1004 ][skypiax2 ][-1, 0, 0] Interface_id=2 > is now STARTED, the Skype client to which we are connected gave us the > correct CURRENTUSERHANDLE (voipserver) > > > > > > > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].interface_id=1 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].X11_display=:101 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].name=skypiax1 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].context=default > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].dialplan=XML > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].destination=3101 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax1 ][-1, 0, 0] i=1 > globals.SKYPIAX_INTERFACES[1].context=default > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].interface_id=2 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].X11_display=:102 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].name=skypiax2 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].context=default > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].dialplan=XML > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].destination=5000 > 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev > 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax2 ][-1, 0, 0] i=2 > globals.SKYPIAX_INTERFACES[2].context=default > 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 > switch_loadable_module_load_file() Successfully Loaded [mod_skypiax] > 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 > switch_loadable_module_process() Adding Endpoint 'skypiax' > 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 > switch_loadable_module_process() Adding API Function 'sk' > 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 > switch_loadable_module_process() Adding API Function 'skypiax' > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] > skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 > ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||USER paolofun6 > PHONE_MOBILE +420775216536||| > > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> > freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 [NOTICE] > switch_channel.c:602 switch_channel_set_name() New Channel sofia/external/ > 07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] > 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel > sofia/external/07771236762 at sipgate.co.uk entering state [received][100] > 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() Remote > SDP: > v=0 > o=root 15141 15141 IN IP4 217.10.66.71 > s=session > c=IN IP4 217.10.66.71 > t=0 0 > m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=30 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:112 G726-32/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 sofia_glue_tech_set_codec() > Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms 160 > samples > 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 sofia_handle_sip_i_state() > (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> CS_INIT > 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [BREAK] > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > Running State Change CS_INIT > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > INIT > 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/external/ > 07771236762 at sipgate.co.uk SOFIA INIT > 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() > (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> > CS_ROUTING > 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [BREAK] > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > INIT going to sleep > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > Running State Change CS_ROUTING > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > ROUTING > 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING > 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 > switch_core_standard_on_routing() sofia/external/07771236762 at sipgate.co.ukStandard ROUTING > 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing 07771236762->00442083324655 in context public > Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing > [public->skype_uri] continue=false > Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) > [skype_uri] destination_number(00442083324655) =~ /^(00442083324655)$/ > break=on-false > Dialplan: sofia/external/07771236762 at sipgate.co.uk Action > bridge(skypiax/skypiax1/xyzTestUK) > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 > switch_core_standard_on_routing() (sofia/external/ > 07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [BREAK] > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > ROUTING going to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > Running State Change CS_EXECUTE > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > EXECUTE > 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() > sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 > switch_core_standard_on_execute() sofia/external/07771236762 at sipgate.co.ukStandard EXECUTE > EXECUTE sofia/external/07771236762 at sipgate.co.ukbridge(skypiax/skypiax1/xyzTestUK) > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 channel_outgoing_channel() > rev 13177[(nil)|37 ][DEBUG_SKYPE 585 ][ ][-1, 0, 0] > globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() rev > 13177[(nil)|37 ][DEBUG_SKYPE 151 ][skypiax1 ][-1, 0, 0] skypiax_codec > SUCCESS > 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New Channel skypiax/skypiax1/xyzTestUK > [0375c668-b4a2-4364-a8c6-0a718d4f00a3] > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() rev > 13177[(nil)|37 ][DEBUG_SKYPE 773 ][skypiax1 ][-1, 0, 0] Calling > Skype, rdest is: xyzTestUK > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > ][skypiax1 ][-1, 0, 0] SENDING: |||SET AGC OFF|||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > ][skypiax1 ][-1, 0, 0] SENDING: |||SET AEC OFF|||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > ][skypiax1 ][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 channel_outgoing_channel() > (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal > skypiax/skypiax1/xyzTestUK [BREAK] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change > CS_INIT > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() > (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal > skypiax/skypiax1/xyzTestUK [BREAK] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() rev > 13177[(nil)|37 ][DEBUG_SKYPE 182 ][skypiax1 ][-1, 0, 0] > skypiax/skypiax1/xyzTestUK CHANNEL INIT > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT going to > sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change > CS_ROUTING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() rev > 13177[(nil)|37 ][DEBUG_SKYPE 257 ][skypiax1 ][-1, 0, 0] > skypiax/skypiax1/xyzTestUK CHANNEL ROUTING > 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 > originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change CS_ROUTING > -> CS_CONSUME_MEDIA > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal > skypiax/skypiax1/xyzTestUK [BREAK] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING going > to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change > CS_CONSUME_MEDIA > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||AGC OFF||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||AEC OFF||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: > |||CALL 455 STATUS UNPLACED||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 0, 0] Skype > MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: UNPLACED,where: > NULL! > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 371 ][skypiax1 ][-1, 3,116] > skype_call: 455 is now UNPLACED > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,116] READING: > |||CALL 455 STATUS ROUTING||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,116] Skype > MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: ROUTING,where: > NULL! > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 365 ][skypiax1 ][-1, 3,117] > skype_call: 455 is now ROUTING > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: > |||CALL 455 FAILUREREASON 7||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype > MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, value: 7,where: > NULL! > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 201 ][skypiax1 ][-1, 3,117] Skype > FAILED on skype_call 455. Let's wait for the FAILED message. > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: > |||CALL 455 VAA_INPUT_STATUS FALSE||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype > MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, value: > FALSE,where: NULL! > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] READING: > |||CALL 455 STATUS FAILED||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype > MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: FAILED,where: > NULL! > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 334 ][skypiax1 ][-1, 3,112] we tried > to call Skype on skype_call 455 and Skype has now FAILED > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 > skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 > ][skypiax1 ][-1, 1,112] skype call ended > 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 > skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK > [CS_CONSUME_MEDIA] [NORMAL_CLEARING] > 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 > switch_channel_perform_hangup() Send signal skypiax/skypiax1/xyzTestUK > [KILL] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 293 ][skypiax1 ][-1, 1,112] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal > skypiax/skypiax1/xyzTestUK [BREAK] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 > switch_ivr_originate() Originate Resulted in Error Cause: 16 > [NORMAL_CLEARING] > 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 audio_bridge_function() > Originate Failed. Cause: NORMAL_CLEARING > 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 audio_bridge_function() > Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 > switch_channel_perform_hangup() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [KILL] > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [BREAK] > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > EXECUTE going to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > Running State Change CS_HANGUP > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > HANGUP > 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel > sofia/external/07771236762 at sipgate.co.uk hanging up, cause: > NORMAL_CLEARING > 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to > INVITE with: 480 > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/external/07771236762 at sipgate.co.ukStandard HANGUP, cause: NORMAL_CLEARING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > HANGUP going to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > Change CS_HANGUP -> CS_REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal sofia/external/ > 07771236762 at sipgate.co.uk [BREAK] > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > Running State Change CS_REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > switch_core_session_reporting_state() (sofia/external/ > 07771236762 at sipgate.co.uk) State REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA > going to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change > CS_HANGUP > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev > 13177[(nil)|37 ][DEBUG_SKYPE 228 ][skypiax1 ][-1, 1,112] hanging up > skype call: 455 > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > ][skypiax1 ][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev > 13177[(nil)|37 ][DEBUG_SKYPE 235 ][skypiax1 ][-1, 1,112] > skypiax/skypiax1/xyzTestUK CHANNEL HANGUP > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard HANGUP, > cause: NORMAL_CLEARING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP going to > sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change > CS_HANGUP -> CS_REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > switch_core_session_signal_state_change() Send signal > skypiax/skypiax1/xyzTestUK [BREAK] > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev > 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] > skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State Change > CS_REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State > REPORTING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard > REPORTING, cause: NORMAL_CLEARING > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State > REPORTING going to sleep > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change > CS_REPORTING -> CS_DESTROY > 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 > switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Locked, > Waiting on external entities > 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 > switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Ended > 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 > switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK > [CS_DESTROY] > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State > DESTROY > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard > DESTROY > 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State > DESTROY going to sleep > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 1,112] READING: > |||ERROR 559 CALL: Action failed||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 91 ][skypiax1 ][-1, 1,112] Skype > got ERROR: |||ERROR||| > 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 skypiax_signaling_read() > rev 13177[(nil)|37 ][DEBUG_SKYPE 93 ][skypiax1 ][-1, 1,110] > skype_call now is DOWN > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 > skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 > ][skypiax1 ][-1, 1,110] skype call ended > 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 > skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 687 > ][skypiax1 ][-1, 1,110] no session > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() sofia/external/ > 07771236762 at sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 > switch_core_session_reporting_state() (sofia/external/ > 07771236762 at sipgate.co.uk) State REPORTING going to sleep > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State > Change CS_REPORTING -> CS_DESTROY > 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 > switch_core_session_thread() Session 1 (sofia/external/ > 07771236762 at sipgate.co.uk) Locked, Waiting on external entities > 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 > switch_core_session_thread() Session 1 (sofia/external/ > 07771236762 at sipgate.co.uk) Ended > 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 > switch_core_session_thread() Close Channel sofia/external/ > 07771236762 at sipgate.co.uk [CS_DESTROY] > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/external/ > 07771236762 at sipgate.co.uk) State DESTROY > 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() > sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() sofia/external/07771236762 at sipgate.co.ukStandard DESTROY > 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/external/ > 07771236762 at sipgate.co.uk) State DESTROY going to sleep > -- > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > 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/ 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/76b8b881/attachment-0002.html From gcd at i.ph Thu Apr 30 19:08:01 2009 From: gcd at i.ph (Nandy Dagondon) Date: Fri, 1 May 2009 10:08:01 +0800 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <23317579.post@talk.nabble.com> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> Message-ID: <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> hi guys, i've installed FreeNas using CF-to-IDE adaptor and SanDisk 128MB CF. it's working fine. but i want to try FS on a 16GB Kingston CF. anyone tried this? if none, i can also settle down for 8GB. pls mention which brand/size works. tks, nandy On Thu, Apr 30, 2009 at 10:46 PM, Fred-145 wrote: > > Thanks guys for the links on CF-to-IDE adaptors. > -- > View this message in context: > http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23317579.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/20090501/e328c3c0/attachment-0002.html From mitch.capper at gmail.com Thu Apr 30 20:31:31 2009 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 30 Apr 2009 23:31:31 -0400 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> Message-ID: You may want to look at the Intel Atom combo machines you can get a 1.6 ghz machine probably for around $100-150 USD in a very small form factor and very powerful. ~Mitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/27dcffc4/attachment-0002.html From brian at freeswitch.org Thu Apr 30 20:40:26 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 30 Apr 2009 22:40:26 -0500 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> Message-ID: I have two intel atom boxes sitting on a shelf above my desk ... works like a charm! /b On Apr 30, 2009, at 10:31 PM, Mitch Capper wrote: > You may want to look at the Intel Atom combo machines you can get a > 1.6 ghz machine probably for around $100-150 USD in a very small > form factor and very powerful. > > ~Mitch Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/d7cfae74/attachment-0002.html From mszlazak at aol.com Thu Apr 30 21:07:48 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Fri, 01 May 2009 00:07:48 -0400 Subject: [Freeswitch-users] Latest SVN update gives Windows Express compiler errors ... Message-ID: <8CB98298639AEA6-280-33C3@webmail-dx08.sysops.aol.com> I'm getting Windows Express compiler errors on the latest svn update to trunk 13213. It looks like the path is wrong to some files. Instead of folder "Debug", it's looking for files in folder "Debug DLL" Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/0074253d/attachment-0002.html From gcd at i.ph Thu Apr 30 21:16:09 2009 From: gcd at i.ph (Nandy Dagondon) Date: Fri, 1 May 2009 12:16:09 +0800 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> Message-ID: <7d0bfd8c0904302116x71e1746es4c23dee52a894eed@mail.gmail.com> rhino used the dual-core atom mobo d945gclf2 but it requires downloading/building the linux r8168 LAN driver. -nandy On Fri, May 1, 2009 at 11:40 AM, Brian West wrote: > I have two intel atom boxes sitting on a shelf above my desk ... works like > a charm! > /b > > On Apr 30, 2009, at 10:31 PM, Mitch Capper wrote: > > You may want to look at the Intel Atom combo machines you can get a 1.6 > ghz machine probably for around $100-150 USD in a very small form factor and > very powerful. > > ~Mitch > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/7cf3c47e/attachment-0002.html From brian at freeswitch.org Thu Apr 30 21:18:49 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 30 Apr 2009 23:18:49 -0500 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <7d0bfd8c0904302116x71e1746es4c23dee52a894eed@mail.gmail.com> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> <7d0bfd8c0904302116x71e1746es4c23dee52a894eed@mail.gmail.com> Message-ID: Sounds like the MSI Wind :P I had to do the same thing! /b On Apr 30, 2009, at 11:16 PM, Nandy Dagondon wrote: > rhino used the dual-core atom mobo d945gclf2 but it requires > downloading/building the linux r8168 LAN driver. > > -nandy Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090430/ef0fabe2/attachment-0002.html From technical at ttnc.co.uk Thu Apr 30 22:50:16 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Fri, 01 May 2009 06:50:16 +0100 Subject: [Freeswitch-users] uuid_displace & FIFO help In-Reply-To: <191c3a030904270519h6f85d391p72ca1500f94cfaa5@mail.gmail.com> References: <49F07EC5.5040504@barakatdesigns.net> <18FCD53D-A0ED-4A5A-80A7-A9C7E1FF3349@freeswitch.org> <49F0840C.7030305@ttnc.co.uk> <191c3a030904231418h6d11e11bp4e84f44ea2abf179@mail.gmail.com> <49F0E3A8.5030400@ttnc.co.uk> <49F16826.5050203@ttnc.co.uk> <6D57020E-7D08-4886-A2BC-6F139E6C1BD6@freeswitch.org> <49F1C880.5040300@ttnc.co.uk> <191c3a030904241859w4bc26e84u3d8640dda76a961e@mail.gmail.com> <49F56F3B.8000906@ttnc.co.uk> <191c3a030904270519h6f85d391p72ca1500f94cfaa5@mail.gmail.com> Message-ID: <49FA8D98.3040900@ttnc.co.uk> Anthony Minessale wrote: > Also is there any way to stop uuid_broadcast as I'd > need to stop it somehow if the destination picks up? > > break all "uuid_broadcast phrase::saynumber,1" doesn't set the 'current_application_response' variable in the same way as "uuid_broadcast playback::filename.wav" does (which my script looks for to know when to move on to the next application). I've attached a patch which sets this variable if it's any use to anyone (I'm not that great at C so I hope it's correct, any comments/improvements are welcome). Thanks again Adnan -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: mod_dptools.patch Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/095a4d23/attachment-0002.pl From gmaruzz at celliax.org Thu Apr 30 23:20:10 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 1 May 2009 08:20:10 +0200 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error In-Reply-To: <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> References: <20090430223701.280500@gmx.net> <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> Message-ID: <7b197bef0904302320t6d025985vc4e912b4373577b1@mail.gmail.com> Have a happy MayDay! I cannot see the whole mail now, it's clipped for my mobile, but it seems the nth bizarry of new alsa config file, that creates an hdmi device even if you do not have one. Try to edit /usr/share/alsa/alsa.conf or any other file in /usr/share/alsa dir and delete any mention of 'hdmi'. If this do not works, please file a jira or write again. Giovanni On 5/1/09, Anthony Minessale wrote: > if you put that info in a jira ticket > > http://jira.freeswitch.org > > and route it to skypeiax , the guy who maintains that module will see it. > > > On Thu, Apr 30, 2009 at 5:37 PM, wrote: > >> >> Hello, >> >> I am trying to get skypiax working, but I am having trouble with the >> sound. >> The calls fail with CALL FAILUREREASON 7 = Sound I/O error and >> I am getting the following error: >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM >> cards.pcm.hdmi >> >> >> I am running centos 5.3 and have followed the installation guide on the >> wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When >> saving >> the configuration on my desktop I have set the sound card to snd_dummy. On >> the server the startup script load snd-dumy like this /sbin/modprobe >> snd-dummy enable=1. >> Below is the output of lsmod and the debug output from FS. It would be >> great if someone could help me fix my problem. >> >> Thank you very much. >> Best wishes, >> Phil >> >> >> >> >> -bash-3.2# lsmod >> Module Size Used by >> snd_dummy 12416 0 >> snd_seq_oss 32832 0 >> snd_seq_midi_event 7744 1 snd_seq_oss >> snd_seq 55200 4 snd_seq_oss,snd_seq_midi_event >> snd_seq_device 7120 1 snd_seq_oss >> snd_pcm_oss 44480 0 >> snd_mixer_oss 16512 1 snd_pcm_oss >> snd_pcm 79624 2 snd_dummy,snd_pcm_oss >> snd_timer 22088 2 snd_seq,snd_pcm >> snd 55976 8 >> snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer >> soundcore 7456 1 snd >> snd_page_alloc 8720 1 snd_pcm >> >> >> >> freeswitch at voipserverServerFreeswitch> load mod_skypiax >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 718 ][none ][-1,-1,-1] >> globals.debug=0 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 720 ][none ][-1,-1,-1] >> globals.debug=8 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 731 ][none ][-1,-1,-1] >> codec-master >> globals.debug=8 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 734 ][none ][-1,-1,-1] >> globals.dialplan=XML >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 740 ][none ][-1,-1,-1] >> globals.context=default >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 743 ][none ][-1,-1,-1] >> globals.codec_string=gsm,ulaw >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 750 ][none ][-1,-1,-1] >> globals.codec_rates_string=8000,16000 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 723 ][none ][-1,-1,-1] >> globals.hold_music= >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 737 ][none ][-1,-1,-1] >> globals.destination=5000 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] >> interface_id=1 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] >> name=skypiax1 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized >> XInitThreads! >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax1 ][-1, 0, 0] CONFIGURING >> interface_id=1 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].destination=3101 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax1 ][-1, 0, 0] >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev >> 13177[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING >> interface_id=1 >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE >> 1407 >> ][skypiax1 ][-1, 0, 0] X Display ':101' opened >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype >> instance found with id #2097454 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 >> ][skypiax1 ][-1, 0, 0] In skypiax_signaling_thread_func: started, >> p=0x2aaab93226f8 >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||OK||| >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||PROTOCOL 7||| >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||CONNSTATUS ONLINE||| >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||CURRENTUSERHANDLE xyzUK||| >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax1 ][-1, 0, 0] Skype >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, >> cuh: >> xyzUK, skype_user: xyzUK! >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||USERSTATUS ONLINE||| >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev >> 13177[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly >> 10 >> seconds to find a running Skype client and connect to its SKYPE API for >> interface_id=1 >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev >> 13177[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running >> Skype client, connected to its SKYPE API for interface_id=1, waiting 60 >> seconds for CURRENTUSERHANDLE==xyzUK >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev >> 13177[(nil)|37 ][WARNINGA 1004 ][skypiax1 ][-1, 0, 0] Interface_id=1 >> is now STARTED, the Skype client to which we are connected gave us the >> correct CURRENTUSERHANDLE (xyzUK) >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] >> interface_id=2 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] >> name=skypiax2 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] Initialized >> XInitThreads! >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax2 ][-1, 0, 0] CONFIGURING >> interface_id=2 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax2 ][-1, 0, 0] >> interface_id=2 >> globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].destination=5000 >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax2 ][-1, 0, 0] >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev >> 13177[(nil)|37 ][WARNINGA 950 ][skypiax2 ][-1, 0, 0] STARTING >> interface_id=2 >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE >> 1407 >> ][skypiax2 ][-1, 0, 0] X Display ':102' opened >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype >> instance found with id #2097454 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 661 >> ][skypiax2 ][-1, 0, 0] In skypiax_signaling_thread_func: started, >> p=0x2aaab9325c18 >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] >> READING: >> |||OK||| >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] >> READING: >> |||PROTOCOL 7||| >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] >> READING: >> |||CONNSTATUS ONLINE||| >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] >> READING: >> |||CURRENTUSERHANDLE voipserver||| >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax2 ][-1, 0, 0] Skype >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, >> cuh: >> voipserver, skype_user: voipserver! >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] >> READING: >> |||USERSTATUS ONLINE||| >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev >> 13177[(nil)|37 ][NOTICA 976 ][skypiax2 ][-1, 0, 0] WAITING roughly >> 10 >> seconds to find a running Skype client and connect to its SKYPE API for >> interface_id=2 >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev >> 13177[(nil)|37 ][NOTICA 986 ][skypiax2 ][-1, 0, 0] Found a running >> Skype client, connected to its SKYPE API for interface_id=2, waiting 60 >> seconds for CURRENTUSERHANDLE==voipserver >> API CALL [load(mod_skypiax)] output: >> +OK >> >> 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev >> 13177[(nil)|37 ][WARNINGA 1004 ][skypiax2 ][-1, 0, 0] Interface_id=2 >> is now STARTED, the Skype client to which we are connected gave us the >> correct CURRENTUSERHANDLE (voipserver) >> >> >> >> >> >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].interface_id=1 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].X11_display=:101 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].name=skypiax1 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].context=default >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].dialplan=XML >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].destination=3101 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax1 ][-1, 0, 0] i=1 >> globals.SKYPIAX_INTERFACES[1].context=default >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].interface_id=2 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].X11_display=:102 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].name=skypiax2 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].context=default >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].dialplan=XML >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].destination=5000 >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax2 ][-1, 0, 0] i=2 >> globals.SKYPIAX_INTERFACES[2].context=default >> 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 >> switch_loadable_module_load_file() Successfully Loaded [mod_skypiax] >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 >> switch_loadable_module_process() Adding Endpoint 'skypiax' >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 >> switch_loadable_module_process() Adding API Function 'sk' >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 >> switch_loadable_module_process() Adding API Function 'skypiax' >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] >> skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 >> ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||USER paolofun6 >> PHONE_MOBILE +420775216536||| >> >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 [NOTICE] >> switch_channel.c:602 switch_channel_set_name() New Channel sofia/external/ >> 07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() >> Channel >> sofia/external/07771236762 at sipgate.co.uk entering state [received][100] >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() Remote >> SDP: >> v=0 >> o=root 15141 15141 IN IP4 217.10.66.71 >> s=session >> c=IN IP4 217.10.66.71 >> t=0 0 >> m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=fmtp:97 mode=30 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:112 G726-32/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 sofia_glue_tech_set_codec() >> Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms 160 >> samples >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp() >> Set 2833 dtmf payload to 101 >> 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 sofia_handle_sip_i_state() >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> CS_INIT >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [BREAK] >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> Running State Change CS_INIT >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> INIT >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/external/ >> 07771236762 at sipgate.co.uk SOFIA INIT >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> >> CS_ROUTING >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [BREAK] >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> INIT going to sleep >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> Running State Change CS_ROUTING >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> ROUTING >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() >> sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 >> switch_core_standard_on_routing() >> sofia/external/07771236762 at sipgate.co.ukStandard ROUTING >> 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing 07771236762->00442083324655 in context public >> Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing >> [public->skype_uri] continue=false >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) >> [skype_uri] destination_number(00442083324655) =~ /^(00442083324655)$/ >> break=on-false >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Action >> bridge(skypiax/skypiax1/xyzTestUK) >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 >> switch_core_standard_on_routing() (sofia/external/ >> 07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [BREAK] >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> ROUTING going to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> Running State Change CS_EXECUTE >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> EXECUTE >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() >> sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 >> switch_core_standard_on_execute() >> sofia/external/07771236762 at sipgate.co.ukStandard EXECUTE >> EXECUTE >> sofia/external/07771236762 at sipgate.co.ukbridge(skypiax/skypiax1/xyzTestUK) >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 channel_outgoing_channel() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 585 ][ ][-1, 0, 0] >> globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 151 ][skypiax1 ][-1, 0, 0] >> skypiax_codec >> SUCCESS >> 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() >> New Channel skypiax/skypiax1/xyzTestUK >> [0375c668-b4a2-4364-a8c6-0a718d4f00a3] >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 773 ][skypiax1 ][-1, 0, 0] Calling >> Skype, rdest is: xyzTestUK >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 >> ][skypiax1 ][-1, 0, 0] SENDING: |||SET AGC OFF|||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 >> ][skypiax1 ][-1, 0, 0] SENDING: |||SET AEC OFF|||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 >> ][skypiax1 ][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 channel_outgoing_channel() >> (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal >> skypiax/skypiax1/xyzTestUK [BREAK] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> Change >> CS_INIT >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() >> (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal >> skypiax/skypiax1/xyzTestUK [BREAK] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 182 ][skypiax1 ][-1, 0, 0] >> skypiax/skypiax1/xyzTestUK CHANNEL INIT >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT going to >> sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> Change >> CS_ROUTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 257 ][skypiax1 ][-1, 0, 0] >> skypiax/skypiax1/xyzTestUK CHANNEL ROUTING >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 >> originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change >> CS_ROUTING >> -> CS_CONSUME_MEDIA >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal >> skypiax/skypiax1/xyzTestUK [BREAK] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING going >> to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> Change >> CS_CONSUME_MEDIA >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||AGC OFF||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||AEC OFF||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] >> READING: >> |||CALL 455 STATUS UNPLACED||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 0, 0] Skype >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >> UNPLACED,where: >> NULL! >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 371 ][skypiax1 ][-1, 3,116] >> skype_call: 455 is now UNPLACED >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,116] >> READING: >> |||CALL 455 STATUS ROUTING||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,116] Skype >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >> ROUTING,where: >> NULL! >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 365 ][skypiax1 ][-1, 3,117] >> skype_call: 455 is now ROUTING >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] >> READING: >> |||CALL 455 FAILUREREASON 7||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype >> MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, value: >> 7,where: >> NULL! >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 201 ][skypiax1 ][-1, 3,117] Skype >> FAILED on skype_call 455. Let's wait for the FAILED message. >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] >> READING: >> |||CALL 455 VAA_INPUT_STATUS FALSE||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype >> MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, value: >> FALSE,where: NULL! >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] >> READING: >> |||CALL 455 STATUS FAILED||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] Skype >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: FAILED,where: >> NULL! >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 >> skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 334 ][skypiax1 ][-1, 3,112] we >> tried >> to call Skype on skype_call 455 and Skype has now FAILED >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 >> ][skypiax1 ][-1, 1,112] skype call ended >> 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 >> skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK >> [CS_CONSUME_MEDIA] [NORMAL_CLEARING] >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 >> switch_channel_perform_hangup() Send signal skypiax/skypiax1/xyzTestUK >> [KILL] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 293 ][skypiax1 ][-1, 1,112] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal >> skypiax/skypiax1/xyzTestUK [BREAK] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 >> switch_ivr_originate() Originate Resulted in Error Cause: 16 >> [NORMAL_CLEARING] >> 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 audio_bridge_function() >> Originate Failed. Cause: NORMAL_CLEARING >> 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 audio_bridge_function() >> Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] >> [NORMAL_CLEARING] >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 >> switch_channel_perform_hangup() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [KILL] >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [BREAK] >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> EXECUTE going to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> Running State Change CS_HANGUP >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> HANGUP >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel >> sofia/external/07771236762 at sipgate.co.uk hanging up, cause: >> NORMAL_CLEARING >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding >> to >> INVITE with: 480 >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() >> sofia/external/07771236762 at sipgate.co.ukStandard HANGUP, cause: >> NORMAL_CLEARING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> HANGUP going to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> Change CS_HANGUP -> CS_REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal sofia/external/ >> 07771236762 at sipgate.co.uk [BREAK] >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> Running State Change CS_REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >> switch_core_session_reporting_state() (sofia/external/ >> 07771236762 at sipgate.co.uk) State REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA >> going to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> Change >> CS_HANGUP >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 228 ][skypiax1 ][-1, 1,112] hanging up >> skype call: 455 >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 >> ][skypiax1 ][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 235 ][skypiax1 ][-1, 1,112] >> skypiax/skypiax1/xyzTestUK CHANNEL HANGUP >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard >> HANGUP, >> cause: NORMAL_CLEARING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP going >> to >> sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change >> CS_HANGUP -> CS_REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> switch_core_session_signal_state_change() Send signal >> skypiax/skypiax1/xyzTestUK [BREAK] >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> Change >> CS_REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State >> REPORTING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 >> switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard >> REPORTING, cause: NORMAL_CLEARING >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State >> REPORTING going to sleep >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change >> CS_REPORTING -> CS_DESTROY >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) >> Locked, >> Waiting on external entities >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Ended >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 >> switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK >> [CS_DESTROY] >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State >> DESTROY >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 >> switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard >> DESTROY >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State >> DESTROY going to sleep >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 1,112] >> READING: >> |||ERROR 559 CALL: Action failed||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 91 ][skypiax1 ][-1, 1,112] Skype >> got ERROR: |||ERROR||| >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 skypiax_signaling_read() >> rev 13177[(nil)|37 ][DEBUG_SKYPE 93 ][skypiax1 ][-1, 1,110] >> skype_call now is DOWN >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 672 >> ][skypiax1 ][-1, 1,110] skype call ended >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE 687 >> ][skypiax1 ][-1, 1,110] no session >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 >> switch_core_standard_on_reporting() sofia/external/ >> 07771236762 at sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 >> switch_core_session_reporting_state() (sofia/external/ >> 07771236762 at sipgate.co.uk) State REPORTING going to sleep >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) State >> Change CS_REPORTING -> CS_DESTROY >> 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 >> switch_core_session_thread() Session 1 (sofia/external/ >> 07771236762 at sipgate.co.uk) Locked, Waiting on external entities >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 >> switch_core_session_thread() Session 1 (sofia/external/ >> 07771236762 at sipgate.co.uk) Ended >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 >> switch_core_session_thread() Close Channel sofia/external/ >> 07771236762 at sipgate.co.uk [CS_DESTROY] >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/external/ >> 07771236762 at sipgate.co.uk) State DESTROY >> 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >> sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 >> switch_core_standard_on_destroy() >> sofia/external/07771236762 at sipgate.co.ukStandard DESTROY >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/external/ >> 07771236762 at sipgate.co.uk) State DESTROY going to sleep >> -- >> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + >> Telefonanschluss f?r nur 17,95 Euro/mtl.!* >> http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> 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/ > > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039