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 <mszlazak at aol.com>

>  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 <msc at freeswitch.org>
> 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<http://pr.atwola.com/promoclk/100126575x1220631247x1201390185/aol?redir=http:%2F%2Fad.doubleclick.net%2Fclk%3B213540506%3B35046329%3Bx>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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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 <ribs at acm.org>
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  <mszlazak at aol.com>




 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 <msc at freeswitch.org>

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








 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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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 <msc at freeswitch.org>
> 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

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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 <msc at freeswitch.org>
> 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



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



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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 <dujinfang at gmail.com> 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 <dujinfang at gmail.com> 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 <mailto:brian at freeswitch.org>
>
> -- Meet us a ClueCon!  http://www.cluecon.com <http://www.cluecon.com/>
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.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
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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 <mszlazak at aol.com>

>  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 <msc at freeswitch.org>
> 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<http://pr.atwola.com/promoclk/100126575x1220631247x1201390185/aol?redir=http:%2F%2Fad.doubleclick.net%2Fclk%3B213540506%3B35046329%3Bx>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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


<action application="set" data="conference_enforce_security=false"/>
<action application="conference" data="3000"/>




On Wed, Apr 1, 2009 at 6:41 AM, Peter P GMX <Prometheus001 at gmx.net> 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 <mailto:brian at freeswitch.org>
> >
> > -- Meet us a ClueCon!  http://www.cluecon.com <http://www.cluecon.com/>
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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  
<intralanman at freeswitch.org> 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
<SIP version=?4.1?>
  <content type=?SIP-INVITE?>
    <INVITE recipient=?bob at alice.com?>
      <data type=?sip-2/0?/>
      <![CDATA[INVITE bob at alice.com SIP 2.0
To: bob at alice.com
From: alice at bob.com
Subject: SIP Rocks
]]>
      </data>
    </INVITE>
  </content>
</SIP>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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 <to%3Acomments 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 <to%3Acomments 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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: <A2B4F48489739E439D8845296A2FD9C408BD6A@sun.Noble.local>

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
<SIP version="4.1">
  <content type="SIP-INVITE">
    <INVITE recipient="bob at alice.com">
      <data type="sip-2/0"/>
      <![CDATA[INVITE bob at alice.com SIP 2.0
To: bob at alice.com
From: alice at bob.com
Subject: SIP Rocks
]]>
      </data>
    </INVITE>
  </content>  
</SIP> 



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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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: <A2B4F48489739E439D8845296A2FD9C408BD6A@sun.Noble.local>
References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C408BD6A@sun.Noble.local>
Message-ID: <FAF91982-245C-4BC7-8848-18EF9449D21D@freeswitch.org>

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



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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 <anthony.minessale at 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
>
>
> <action application="set" data="conference_enforce_security=false"/>
> <action application="conference" data="3000"/>
>
>
FYI this is now in the wiki:
http://wiki.freeswitch.org/wiki/Channel_Variables#conference_enforce_security

-MC
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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
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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 <anthony.minessale at 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
> <SIP version=?4.1?>
>   <content type=?SIP-INVITE?>
>     <INVITE recipient=?bob at alice.com?>
>       <data type=?sip-2/0?/>
>       <![CDATA[INVITE bob at alice.com SIP 2.0
> To: bob at alice.com
> From: alice at bob.com
> Subject: SIP Rocks
> ]]>
>       </data>
>     </INVITE>
>   </content>
> </SIP>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 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
>
>
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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: <A57B3E2B-E6DA-41D4-8A3F-DC81AA1B1C1F@freeswitch.org>

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



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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 <raul at etellicom.com>
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 <msc at freeswitch.org>
> 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


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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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



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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 <brian at freeswitch.org>
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
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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: <DCB42C10-E79C-4B23-B5E5-D1325638F48A@freeswitch.org>


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



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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 <anthony.minessale at gmail.com>
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  <mszlazak at aol.com>




 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 <msc at freeswitch.org>

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

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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
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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 <brian at freeswitch.org>
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
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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: <A2B4F48489739E439D8845296A2FD9C408BD6A@sun.Noble.local>
References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C408BD6A@sun.Noble.local>
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
<SIP version="4.1">
  <content type="SIP-INVITE">
    <INVITE recipient="bob at alice.com">
      <data type="sip-2/0"/>
      <![CDATA[INVITE bob at alice.com SIP 2.0
To: bob at alice.com
From: alice at bob.com
Subject: SIP Rocks
]]>
      </data>
    </INVITE>
  </content>  
</SIP> 



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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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: <DCB42C10-E79C-4B23-B5E5-D1325638F48A@freeswitch.org>
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>
	<DCB42C10-E79C-4B23-B5E5-D1325638F48A@freeswitch.org>
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 <brian at freeswitch.org>
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
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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 <mszlazak at aol.com>

>  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/<insert-your-complain-here>: 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
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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>
	<A2B4F48489739E439D8845296A2FD9C408BD6A@sun.Noble.local>
	<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
> <SIP version=?4.1?>
>   <content type=?SIP-INVITE?>
>     <INVITE recipient=?bob at alice.com?>
>       <data type=?sip-2/0?/>
>       <![CDATA[INVITE bob at alice.com SIP 2.0
> To: bob at alice.com
> From: alice at bob.com
> Subject: SIP Rocks
> ]]>
>       </data>
>     </INVITE>
>   </content>  
> </SIP> 
> 
> 
> 
> -- 
> 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 <mszlazak at 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 <raul at etellicom.com>
> 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 <msc at freeswitch.org>
>
> > 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
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
> Freeswitch-users at 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<http://pr.atwola.com/promoclk/100126575x1220631247x1201390185/aol?redir=http:%2F%2Fad.doubleclick.net%2Fclk%3B213540506%3B35046329%3Bx>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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 <anthony.minessale at 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
>
>
> <action application="set" data="conference_enforce_security=false"/>
> <action application="conference" data="3000"/>
>
>
> 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

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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
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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
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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 <grevenx at me.com>

> 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
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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 <carlos.talbot at 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.
>

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
>
>
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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 <timr at asteriasgi.com> 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
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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: <b39273530904011156w333ccde2qc008482c02083ecd@mail.gmail.com>

2009/4/1 Michael Collins <msc at freeswitch.org>

> 2009/4/1 Carlos Talbot <carlos.talbot at 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.
>>
>
> 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
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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 <timr at asteriasgi.com> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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 <msc at freeswitch.org>

> 2009/4/1 Carlos Talbot <carlos.talbot at 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.
>>
>
> 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
>
>
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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 <carlos.talbot at 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 <msc at freeswitch.org>
>
>> 2009/4/1 Carlos Talbot <carlos.talbot at 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.
>>>
>>
>> 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
>
>
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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: <edf696df0904011327g220582cav9357afee5e04b6a1@mail.gmail.com>

where can see and buy that router?

2009/4/1 Carlos Talbot <carlos.talbot at 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
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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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 <anthony.minessale at gmail.com>
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  <mszlazak at 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 <raul at etellicom.com>

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 <msc at freeswitch.org>

> 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





_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at 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



 

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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 <intralanman at freeswitch.org>
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



 

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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: <FAF91982-245C-4BC7-8848-18EF9449D21D@freeswitch.org>
References: <191c3a030904010831x7708544dkd0b7f523486ba380@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C408BD6A@sun.Noble.local>
	<FAF91982-245C-4BC7-8848-18EF9449D21D@freeswitch.org>
Message-ID: <AC99CB4B-75D0-432C-A470-522931291D90@3c.co.uk>

Here's a sample SIP/SMTP INVITE (responses omitted for clarity)
MAIL FROM: <dave at 3c.co.uk>
RCPT TO: <marina at 3c.co.uk>
DATA
Call me
.

--Dave

Sent from my iPhone

On 1 Apr 2009, at 09:15, Brian West <brian at freeswitch.org> 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
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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

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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: <AEBAC917-0855-4D11-91E6-B5CC68508BA3@freeswitch.org>

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



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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: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>


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/
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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:

<extension name="9999">
  <condition field="destination_number" expression="^9999$">
    <action application="verbose_events" data="true"/>
    <action application="socket" data="192.168.251.1:8384 async full"/>
  </condition>
</extension>

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: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>

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
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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: <b39273530904011415q5eb03337x5fdf35a790ec5783@mail.gmail.com>

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 <stevecrozz at 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 <carlos.talbot at 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 <msc at freeswitch.org>
>>
>>> 2009/4/1 Carlos Talbot <carlos.talbot at 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.
>>>>
>>>
>>> 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
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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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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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: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
References: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
Message-ID: <87f2f3b90904011418j371e1363m4ec1def2e9ba2818@mail.gmail.com>

2009/4/1 Stromin Normin <stormin.normin at hotmail.co.uk>

>  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
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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: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
References: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
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 <stormin.normin at hotmail.co.uk>

>  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<http://extras.uk.msn.com/internet-explorer-8/?ocid=T010MSN07A0716U>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>
References: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>
Message-ID: <87f2f3b90904011422i32e2517dhd5cfc9414468c08@mail.gmail.com>

Rupa,

Thanks for adding to the project! Well done.
-MC

2009/4/1 Rupa Schomaker <rupa at rupa.com>

> 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
>
>
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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 <toofics at gmail.com> 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:
>
> <extension name="9999">
>  <condition field="destination_number" expression="^9999$">
>    <action application="verbose_events" data="true"/>
>    <action application="socket" data="192.168.251.1:8384 async full"/>
>  </condition>
> </extension>
>
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>
References: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>
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 <rupa at rupa.com>

> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>
	<191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com>
Message-ID: <b39273530904011431r2e473c6fgfeff7ca3b3df8bc9@mail.gmail.com>

2009/4/1 Anthony Minessale <anthony.minessale at 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 ;)
>

haha!  I didn't even think of that.

-- 
-Rupa
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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: <b39273530904011431r2e473c6fgfeff7ca3b3df8bc9@mail.gmail.com>
References: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>
	<191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com>
	<b39273530904011431r2e473c6fgfeff7ca3b3df8bc9@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 <anthony.minessale at 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 ;)
>
> 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

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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: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
References: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
Message-ID: <BLU146-W145F7C8A442F95403B1817D58B0@phx.gbl>


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
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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: <BLU146-W145F7C8A442F95403B1817D58B0@phx.gbl>
References: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
	<BLU146-W145F7C8A442F95403B1817D58B0@phx.gbl>
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 <stormin.normin at hotmail.co.uk>:
> 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: <b39273530904011529l207e0252obeaed1f1bdcf4547@mail.gmail.com>

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:
        <action application="set" "enable_file_write_buffering=false"/>

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
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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: <BLU146-W145F7C8A442F95403B1817D58B0@phx.gbl>
References: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
	<BLU146-W145F7C8A442F95403B1817D58B0@phx.gbl>
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: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
	<BLU146-W145F7C8A442F95403B1817D58B0@phx.gbl>
	<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 <thorhs at basis.is>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
>
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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: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
	<BLU146-W145F7C8A442F95403B1817D58B0@phx.gbl>
	<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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>.
Contact: <sip:999999999 at 2.2.2.2>.
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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>.
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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>;tag=ceKFmNU84B90c.
Contact: <sip:999999999 at 2.2.2.2>.
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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>.
Contact: <sip:999999999 at 2.2.2.2>.
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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>.
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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
To: <sip:666666666 at 3.3.3.3>.
Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
CSeq: 113193247 INVITE.
Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
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" <sip: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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
To: <sip:666666666 at 3.3.3.3>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
CSeq: 103 INVITE.
Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>.
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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
CSeq: 113193247 ACK.
Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
CSeq: 103 INVITE.
Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
Contact: <sip:999999999 at 2.2.2.2>.
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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
To: <sip:666666666 at 1.1.1.1>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
CSeq: 113193248 BYE.
Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
To: <sip:666666666 at 3.3.3.3>;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: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
	<BLU146-W145F7C8A442F95403B1817D58B0@phx.gbl>
	<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 <stormin.normin at hotmail.co.uk>:
> > 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: <B8963D4E-70A4-405F-80AB-FF325B4FD8F0@freeswitch.org>

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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=ceKFmNU84B90c.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193247 INVITE.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip: 
> 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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
> CSeq: 103 INVITE.
> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
> 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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193247 ACK.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
> CSeq: 103 INVITE.
> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193248 BYE.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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



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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: <B8963D4E-70A4-405F-80AB-FF325B4FD8F0@freeswitch.org>
References: <8b3b7acc0904011536k27fc2ce1x8fc119d0fa54b6d@mail.gmail.com>
	<B8963D4E-70A4-405F-80AB-FF325B4FD8F0@freeswitch.org>
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 <brian at freeswitch.org>:
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=ceKFmNU84B90c.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193247 INVITE.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip: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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
> CSeq: 103 INVITE.
> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193247 ACK.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
> CSeq: 103 INVITE.
> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193248 BYE.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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
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> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=ceKFmNU84B90c.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193247 INVITE.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip: 
> 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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
> CSeq: 103 INVITE.
> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
> 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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193247 ACK.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
> CSeq: 103 INVITE.
> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193248 BYE.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip: 
> 559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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



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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 <brian at 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=ceKFmNU84B90c.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193247 INVITE.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip: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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
> CSeq: 103 INVITE.
> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193247 ACK.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
> CSeq: 103 INVITE.
> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193248 BYE.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <B0951BEF-C29C-4AB4-9AF5-2A02CD760901@freeswitch.org>

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



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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 <elhodred at gmail.com> 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: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
	<BLU146-W145F7C8A442F95403B1817D58B0@phx.gbl>
	<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
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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: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>
	<191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com>
	<b39273530904011431r2e473c6fgfeff7ca3b3df8bc9@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
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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 <toofics at gmail.com>
> 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:
>         
>         <extension name="9999">
>          <condition field="destination_number" expression="^9999$">
>            <action application="verbose_events" data="true"/>
>            <action application="socket" data="192.168.251.1:8384 async
>         full"/>
>          </condition>
>         </extension>
>         
>         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: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>
	<191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com>
	<b39273530904011431r2e473c6fgfeff7ca3b3df8bc9@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



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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 <carlos.talbot at 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
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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>
	<B8963D4E-70A4-405F-80AB-FF325B4FD8F0@freeswitch.org>
	<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 <elhodred at gmail.com> 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 <brian at freeswitch.org>:
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>.
>> Contact: <sip:999999999 at 2.2.2.2>.
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>.
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=ceKFmNU84B90c.
>> Contact: <sip:999999999 at 2.2.2.2>.
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>.
>> Contact: <sip:999999999 at 2.2.2.2>.
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>.
>> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>.
>> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
>> CSeq: 113193247 INVITE.
>> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
>> 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" <sip: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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 103 INVITE.
>> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>.
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
>> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
>> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
>> CSeq: 113193247 ACK.
>> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
>> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
>> CSeq: 103 INVITE.
>> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
>> Contact: <sip:999999999 at 2.2.2.2>.
>> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
>> To: <sip:666666666 at 1.1.1.1>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
>> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
>> CSeq: 113193248 BYE.
>> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
>> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
>> To: <sip:666666666 at 3.3.3.3>;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
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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
>   
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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

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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



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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
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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: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
References: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
Message-ID: <BLU146-W37F4C7005A08189456633ED5880@phx.gbl>


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.

"
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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
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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



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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: <b39273530904020537k1f7d3b4esa600b5651123d49@mail.gmail.com>

Update the to the latest.  I've added more channel vars:

eg:

after doing:

        <action application="lcr" data="12148267722 default2"/>
(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) <yivzhenko at mksat.net>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
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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: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>	<191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com>	<b39273530904011431r2e473c6fgfeff7ca3b3df8bc9@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:
        <action application="set" data="ignore=${memcache(set
freeswitch:test 'This is a test')}"/> <!-- Memcache test -->
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 <mailto:brian at freeswitch.org>
>
> -- Meet us a ClueCon!  http://www.cluecon.com <http://www.cluecon.com/>
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.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:

     <param name="accept-blind-reg" value="true"/>
     <param name="accept-blind-auth" value="true"/>

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: <b39273530904011413q4528a0d7t5b33cdfe791977c7@mail.gmail.com>
	<191c3a030904011424x70605d6dn4d11640fb8547aee@mail.gmail.com>
	<b39273530904011431r2e473c6fgfeff7ca3b3df8bc9@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: <b39273530904020613m22c30b08ved3ecff50b5d8ab9@mail.gmail.com>

On Thu, Apr 2, 2009 at 8:08 AM, Peter P GMX <Prometheus001 at gmx.net> 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
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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: <b39273530904011529l207e0252obeaed1f1bdcf4547@mail.gmail.com>
References: <b39273530904011529l207e0252obeaed1f1bdcf4547@mail.gmail.com>
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 <rupa at rupa.com>

> 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:
>         <action application="set" "enable_file_write_buffering=false"/>
>
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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: <b39273530904011529l207e0252obeaed1f1bdcf4547@mail.gmail.com>
	<191c3a030904020651ideeb958v77be91490a6ff38f@mail.gmail.com>
Message-ID: <b39273530904020705u4f78349cmb977ddccc82a7060@mail.gmail.com>

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. <shrug/>

I'll go the local stream route for now....

-- 
-Rupa
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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).
<list name="domains" default="deny">
<node type="allow" domain="$${domain}"/>
<node type="allow" cidr="123.123.123.123/32"/>
<node type="allow" cidr="124.124.124.1/24"/>
</list>


Best regards
Peter

Leon de Rooij schrieb:
> Hi,
>
> You can blindly accept registrations and / or authentication messages  
> with these parameters in a sip profile:
>
>      <param name="accept-blind-reg" value="true"/>
>      <param name="accept-blind-auth" value="true"/>
>
> 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: <E460536D-167F-4F7D-8775-2FECC531C5B5@jerris.com>

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?
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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
<param name="accept-blind-reg" value="true"/>
<param name="accept-blind-auth" value="true"/>


Regards,
Bipin
www.xbipin.com
+971-55-9270058

-------- Original Message  --------
Subject: Re: [Freeswitch-users] upper registration in FS?
From: Peter P GMX <Prometheus001 at gmx.net>
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).
> <list name="domains" default="deny">
> <node type="allow" domain="$${domain}"/>
> <node type="allow" cidr="123.123.123.123/32"/>
> <node type="allow" cidr="124.124.124.1/24"/>
> </list>
> 
> 
> Best regards
> Peter
> 
> Leon de Rooij schrieb:
>> Hi,
>>
>> You can blindly accept registrations and / or authentication messages  
>> with these parameters in a sip profile:
>>
>>      <param name="accept-blind-reg" value="true"/>
>>      <param name="accept-blind-auth" value="true"/>
>>
>> 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_<profile name>.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
 
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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
> <param name="accept-blind-reg" value="true"/>
> <param name="accept-blind-auth" value="true"/>
>
>
> Regards,
> Bipin
> www.xbipin.com
> +971-55-9270058

Brian West
brian at freeswitch.org

-- Meet us a ClueCon!  http://www.cluecon.com



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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_<profile name>.db, core.db ?
>
> Regards
> Richard Lamkin
>
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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>

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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.

<action application="eval" data="${originate(sofia/foo/a at b.com xyz)}"/>
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 <cstomi.levlist at gmail.com>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.
>
> <action application="eval" data="${originate(sofia/foo/a at b.com xyz)}"/>
> 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
>
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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>
	<B8963D4E-70A4-405F-80AB-FF325B4FD8F0@freeswitch.org>
	<8b3b7acc0904011641v2c7e3768ne72e06623ed064f2@mail.gmail.com>
	<2d9149cd0904012309h5428ece5qce9a287b0972dd81@mail.gmail.com>
Message-ID: <E0E8537F-E4C3-4096-A5C8-729FF0EC2A38@freeswitch.org>

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



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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 <cstomi.levlist at gmail.com>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.
>
> <action application="eval" data="${originate(sofia/foo/a at b.com xyz)}"/>
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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 <mailto:brian at freeswitch.org>
> 
> -- Meet us a ClueCon!  http://www.cluecon.com <http://www.cluecon.com/>
> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.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



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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: <sip:0715aaaaaa at xx.xx.xxx.xxx>.
instead of
To: <sip:0715aaaaaa at 62.65.128.62>.

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: <sip:0049xxxxxxxxxxx at 62.65.128.62>;tag=8c977d2613672832fd9d03e9.
To: <sip:0715aaaaaa at 62.65.128.62>.
Call-ID: 8c977d261329cd80fd9d03d6 at 62.65.128.61.
CSeq: 2 INVITE.
User-agent: Netstream VoIP Gateway.
Contact: <sip:0049xxxxxxxxxxx at 62.65.128.62:5060>.
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 <mailto:brian at freeswitch.org>
> 
> -- Meet us a ClueCon!  http://www.cluecon.com <http://www.cluecon.com/>
> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.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



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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: <EAF44032-09EF-4D7C-B25E-B31C1CD281F4@freeswitch.org>

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



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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 <Prometheus001 at gmx.net> 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: <sip:0715aaaaaa at xx.xx.xxx.xxx>.
> instead of
> To: <sip:0715aaaaaa at 62.65.128.62 <sip%3A0715aaaaaa at 62.65.128.62>>.
>
> 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: <sip:0049xxxxxxxxxxx at 62.65.128.62<sip%3A0049xxxxxxxxxxx at 62.65.128.62>
> >;tag=8c977d2613672832fd9d03e9.
> To: <sip:0715aaaaaa at 62.65.128.62 <sip%3A0715aaaaaa at 62.65.128.62>>.
> Call-ID: 8c977d261329cd80fd9d03d6 at 62.65.128.61.
> CSeq: 2 INVITE.
> User-agent: Netstream VoIP Gateway.
> Contact: <sip:0049xxxxxxxxxxx at 62.65.128.62:5060>.
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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 <ceino.no at gmail.com> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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: <EAF44032-09EF-4D7C-B25E-B31C1CD281F4@freeswitch.org>
References: <49D51AEE.7010904@gmx.net>
	<EAF44032-09EF-4D7C-B25E-B31C1CD281F4@freeswitch.org>
Message-ID: <49D52142.7040401@gmx.net>

My ACL contains:
    <list name="domains" default="deny">
      <node type="allow" domain="$${domain}"/>
      <node type="allow" cidr="62.65.128.62/32"/>
    </list>

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 <mailto: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 <mailto:brian at freeswitch.org>
>
> -- Meet us a ClueCon!  http://www.cluecon.com <http://www.cluecon.com/>
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.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>
	<EAF44032-09EF-4D7C-B25E-B31C1CD281F4@freeswitch.org>
	<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 <Prometheus001 at gmx.net> wrote:

> My ACL contains:
>    <list name="domains" default="deny">
>      <node type="allow" domain="$${domain}"/>
>      <node type="allow" cidr="62.65.128.62/32"/>
>    </list>
>
> 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 <mailto: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 <mailto:brian at freeswitch.org>
> >
> > -- Meet us a ClueCon!  http://www.cluecon.com <http://www.cluecon.com/>
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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>	<EAF44032-09EF-4D7C-B25E-B31C1CD281F4@freeswitch.org>	<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 <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     My ACL contains:
>        <list name="domains" default="deny">
>          <node type="allow" domain="$${domain}"/>
>          <node type="allow" cidr="62.65.128.62/32
>     <http://62.65.128.62/32>"/>
>        </list>
>
>     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
>     <mailto:anonymous at 62.65.128.62> <mailto:anonymous at 62.65.128.62
>     <mailto: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 <mailto:brian at freeswitch.org>
>     <mailto:brian at freeswitch.org <mailto:brian at freeswitch.org>>
>     >
>     > -- Meet us a ClueCon!  http://www.cluecon.com
>     <http://www.cluecon.com/>
>     >
>     >
>     >
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > Freeswitch-users mailing list
>     > Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 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 <solko at gcdf.pl> 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 <mailto:brian at freeswitch.org>
> >
> > -- Meet us a ClueCon!  http://www.cluecon.com <http://www.cluecon.com/>
> >
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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
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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 <bipin at xbipin.com> 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.

<condition field="network_addr" expression="^192.168.140.5$">
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 <solko at gcdf.pl> 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 <mailto:brian at freeswitch.org>
>     <mailto:brian at freeswitch.org <mailto:brian at freeswitch.org>>
>     >
>     > -- Meet us a ClueCon!  http://www.cluecon.com
>     <http://www.cluecon.com/>
>     >
>     >
>     >
>     >
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > Freeswitch-users mailing list
>     > Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
> 
> 
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> 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 <brian at 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=ceKFmNU84B90c.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193247 INVITE.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip: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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
> CSeq: 103 INVITE.
> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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: <sip:666666666 at 3.3.3.3:5060>.
> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193247 ACK.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Call-ID: 271c0dad41cc80456b8de2133dc80b2e at 1.1.1.1.
> CSeq: 103 INVITE.
> Contact: <sip:mod_sofia at 1.1.1.1:5060;transport=udp>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;tag=DQc8Ngcc2mZKr.
> Contact: <sip:999999999 at 2.2.2.2>.
> 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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:999999999 at 1.1.1.1>;tag=as26208773.
> To: <sip:666666666 at 1.1.1.1>;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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;tag=731C8E54-1862.
> Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7.
> CSeq: 113193248 BYE.
> Contact: <sip:gw+primus at 1.1.1.1:5060;transport=udp>.
> 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" <sip:559066555 at 3.3.3.3;transport=udp>;tag=e050QBXFZXN6K.
> To: <sip:666666666 at 3.3.3.3>;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 <jason at jasonjgw.net>:
> Alfonso Pinto <elhodred at gmail.com> 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:

  <gateway name="429956">
    <param name="username" value="429956"/>
    <param name="password" value="429956"/>
    <param name="proxy" value="10.0.0.248:5059"/>
    <param name="register" value="false"/>
  </gateway>

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: <B015489E-1C61-4249-8749-C13955A2434F@freeswitch.org>
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 <http://www.cluecon.com/> 





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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: <b39273530904020537k1f7d3b4esa600b5651123d49@mail.gmail.com>
References: <200904021322.05690.yivzhenko@mksat.net>
	<b39273530904020537k1f7d3b4esa600b5651123d49@mail.gmail.com>
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:
>
>         <action application="lcr" data="12148267722 default2"/>
> (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) 
<yivzhenko at mksat.net>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 <solko at gcdf.pl> 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:
>
>  <gateway name="429956">
>    <param name="username" value="429956"/>
>    <param name="password" value="429956"/>
>    <param name="proxy" value="10.0.0.248:5059"/>
>    <param name="register" value="false"/>
>  </gateway>
>
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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 <ashley.ohq at 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
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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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" <rupa at rupa.com> 
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. <shrug/> 

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 
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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: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
References: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
Message-ID: <BLU146-W9C00B8CE6620A566C35C3D5890@phx.gbl>


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
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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: <BLU146-W9C00B8CE6620A566C35C3D5890@phx.gbl>
References: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
	<BLU146-W9C00B8CE6620A566C35C3D5890@phx.gbl>
Message-ID: <F4E1561B-E448-4C82-A90B-D312C8AB5986@freeswitch.org>

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



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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 <lele at windmill.it> 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
>
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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: <F4E1561B-E448-4C82-A90B-D312C8AB5986@freeswitch.org>
References: <BLU146-W129C1DF22F10B9677006D0D58B0@phx.gbl>
	<BLU146-W9C00B8CE6620A566C35C3D5890@phx.gbl>
	<F4E1561B-E448-4C82-A90B-D312C8AB5986@freeswitch.org>
Message-ID: <87f2f3b90904030930r2b82a5c3oa9c558b4c5f7052e@mail.gmail.com>

On Fri, Apr 3, 2009 at 7:11 AM, Brian West <brian at freeswitch.org> wrote:

> Did it sound more like a machine gun?
> /b
>
>
Comfort noise for General Douglas McArthur I guess...
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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



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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"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as05dbf888
[Apr  3 11:25:12] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as5ab1ec7b
[Apr  3 11:25:44] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as5ab1ec7b
[Apr  3 11:36:17] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as5c4625af
[Apr  3 11:55:22] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as4d32ad06
[Apr  3 11:55:54] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as4d32ad06
[Apr  3 11:55:56] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as324c491b
[Apr  3 12:00:19] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as4ab90c05
[Apr  3 12:14:43] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as3edfecbb
[Apr  3 12:23:38] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as305dbb2e
[Apr  3 12:32:14] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as5bf0ab42
[Apr  3 12:49:12] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as7f56ad67
[Apr  3 12:52:21] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as0d5d32e0
[Apr  3 13:10:09] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as1b806860
[Apr  3 13:17:46] NOTICE[16617] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as487f8ecb
[Apr  3 13:29:56] NOTICE[16920] chan_sip.c: Failed to authenticate user
"MeucciSolutions"
<sip:MeucciSolutions at 66.96.218.5<sip%3AMeucciSolutions at 66.96.218.5>
>;tag=as613a9814


On Fri, Apr 3, 2009 at 1:04 PM, Brian West <brian at freeswitch.org> 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
>
>
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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 <mailto:brian at freeswitch.org>
> 
> -- Meet us a ClueCon!  http://www.cluecon.com <http://www.cluecon.com/>
> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.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 <gkuri at ieee.org> 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 <mailto:brian at freeswitch.org>
> >
> > -- Meet us a ClueCon!  http://www.cluecon.com <http://www.cluecon.com/>
> >
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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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



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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 <codecomplete at free.fr> 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
>
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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 <ashley.ohq at 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
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 <ashley.ohq at 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
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at 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
>
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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: <DE3A9C96-F257-4F60-90E3-E5EFC1150310@me.com>

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 <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 <ashley.ohq at 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
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at 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

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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: <DE3A9C96-F257-4F60-90E3-E5EFC1150310@me.com>
References: <150a3aa50904020008p7366d35fw4cb513f7d6ed4aa0@mail.gmail.com>
	<2d9149cd0904031442g7f678ba8k743af073ff4808cd@mail.gmail.com>
	<247f8100904031530g21346996nce4bf0d063526cf7@mail.gmail.com>
	<DE3A9C96-F257-4F60-90E3-E5EFC1150310@me.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 <grevenx at me.com> 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
>> <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 <ashley.ohq at 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
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at 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 <dujinfang at gmail.com> 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



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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 <pablosaro at gmail.com> 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 <pablosaro at gmail.com>
> 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
>
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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 
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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: <CA470FC8-103A-4CB9-95B3-2EB7850079F0@gmail.com>


On Apr 4, 2009, at 7:53 AM, Jason White wrote:

> dujinfang <dujinfang at gmail.com> 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: <C2AF9D22-9F5D-410D-AA0D-EBCAFEBADF3C@gmail.com>


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

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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 <pablosaro at gmail.com> 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: <C2AF9D22-9F5D-410D-AA0D-EBCAFEBADF3C@gmail.com>
References: <25E56D67-6664-4C92-BB93-10BD03D7956B@gmail.com>
	<20090403235300.GA10045@jdc.jasonjgw.net>
	<281EADE3-5318-4C31-8539-310B21DCF4AE@freeswitch.org>
	<C2AF9D22-9F5D-410D-AA0D-EBCAFEBADF3C@gmail.com>
Message-ID: <2d9149cd0904040032n44aa9314v71052dfc2863d0c2@mail.gmail.com>

On Sat, Apr 4, 2009 at 2:41 AM, dujinfang <dujinfang at gmail.com> 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>
	<C2AF9D22-9F5D-410D-AA0D-EBCAFEBADF3C@gmail.com>
	<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>
	<C2AF9D22-9F5D-410D-AA0D-EBCAFEBADF3C@gmail.com>
	<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>
	<C2AF9D22-9F5D-410D-AA0D-EBCAFEBADF3C@gmail.com>
	<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 <dave at 3c.co.uk> 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: <B015489E-1C61-4249-8749-C13955A2434F@freeswitch.org>
Message-ID: <F9EE04D87C3E4946B70FC05485359C67@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 <http://www.cluecon.com/> 





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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: <b39273530903271322j7ddcb0dag54125656d6820889@mail.gmail.com>
References: <49CB8D3D.7050202@ewetel.de>	<3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca>	<49CB98E1.8080705@ewetel.de>	<F71EF941-7E01-4A66-8BFA-6BCF0B5282BD@freeswitch.org>	<49CB9E0C.4030300@ewetel.de>
	<49CCEC15.8010500@ewetel.de>
	<b39273530903271322j7ddcb0dag54125656d6820889@mail.gmail.com>
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)
 
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dnrmpXDLVOnWtjwFKMoVw48=
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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>	<C2AF9D22-9F5D-410D-AA0D-EBCAFEBADF3C@gmail.com>	<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>	<C2AF9D22-9F5D-410D-AA0D-EBCAFEBADF3C@gmail.com>	<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 <dave at 3c.co.uk> 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: <F9EE04D87C3E4946B70FC05485359C67@wsandy>
References: <F9EE04D87C3E4946B70FC05485359C67@wsandy>
Message-ID: <F4014787-4EC5-4C72-8809-4105A519B004@freeswitch.org>

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



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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<your_run_level>.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>
	<C2AF9D22-9F5D-410D-AA0D-EBCAFEBADF3C@gmail.com>
	<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: <c564ccb60904060704n9a04dd6n4ae6d542203949b4@mail.gmail.com>

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
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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: <c564ccb60904060704n9a04dd6n4ae6d542203949b4@mail.gmail.com>
References: <c564ccb60904060704n9a04dd6n4ae6d542203949b4@mail.gmail.com>
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



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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: <c564ccb60904060704n9a04dd6n4ae6d542203949b4@mail.gmail.com>
References: <c564ccb60904060704n9a04dd6n4ae6d542203949b4@mail.gmail.com>
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
> 
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Version: GnuPG v1.4.7 (MingW32)

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6vsgh6Ha34/gdg6iDEEEOR0=
=2H4m
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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>
	<c564ccb60904060704n9a04dd6n4ae6d542203949b4@mail.gmail.com>
	<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-----

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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>
	<c564ccb60904060704n9a04dd6n4ae6d542203949b4@mail.gmail.com>
	<49DA1265.4