From marketing at cluecon.com  Fri Jul  1 00:18:29 2011
From: marketing at cluecon.com (marketing at cluecon.com)
Date: Thu, 30 Jun 2011 20:18:29 +0000
Subject: [Freeswitch-dev] Join Us For A ClueCon Party!
Message-ID: <00000130e233e31e-a6415c1b-c879-40f9-8be2-85a1992f9ec5-000000@email.amazonses.com>

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From jan.berger at video24.no  Sat Jul  2 06:13:01 2011
From: jan.berger at video24.no (Jan Berger)
Date: Sat, 2 Jul 2011 04:13:01 +0200
Subject: [Freeswitch-dev] SpiderMonkey 1.8.5 vs V8
Message-ID: <3CFD4D12EBC74C48A6F7D1D234ED34A0@dell9400>

Hi,

 

I just compiled SpiderMonkey 1.8.5 and V8 into the same app to simulate
behaviour of a VXML browser. The purpose was to see if it was worth
switching from V8 to SpiderMonkey on the JavaScript cluster we use for
VXML/CCXML. The idea was that we maybe could upgrade FS and re-use this work
in FS rather than going through the pain of maintaining a separate
JavaScript engine. 

 

SpiderMonkey 1-8-5 execute one of the test scripts ca 4000 times per second.
V8 managed 60,000 executions per second on the same script.

 

It's a arger difference than I expected based on some of the benchmarks I
have seen, but well speed alone is not everything. V8 do not support Sparc
:-( etc.

 

Jan

 

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From prashant.lamba at gmail.com  Sat Jul  2 15:16:52 2011
From: prashant.lamba at gmail.com (Prashant Lamba)
Date: Sat, 2 Jul 2011 16:46:52 +0530
Subject: [Freeswitch-dev] Removing SDP from 183/180
In-Reply-To: <4E0B1566.8000903@supermedia.pl>
References: <4DDF5CEE.5060505@supermedia.pl>
	<BANLkTik2OqWE+4BPRD08fBXrnCxwY1cfZg@mail.gmail.com>
	<4E0B1566.8000903@supermedia.pl>
Message-ID: <CAPHrvXFY1rLUeicg_Cn7Q2eMFNLPGds1uNejxaknCF+gcqvUTQ@mail.gmail.com>

    * Be certain that you have inbound-bypass-media turned off "
And I want to have it turned on.

If you need to not send 183, then you MUST have bypass_media enabled. I have
tested it and it works. Just so you know, if FS is not doing media, that
means its not sending 183, hence someone (other app) needs to do it. Hence
bypass_media need to be true

Prashant Lamba
Phonologies





On Wed, Jun 29, 2011 at 5:37 PM, Artur Cichocki <acichocki at supermedia.pl>wrote:

> W dniu 27.06.2011 21:58, Anthony Minessale pisze:
> > this functionality is already implemented with
> {ignore_early_media=ring_ready}
> >
> I tried all combinations of ignore_early_media with bybass-media, it
> didn't work.
>
> According to http://wiki.freeswitch.org/wiki/Early_Media:
> " Troubleshooting
>
>     * Be certain that you have inbound-bypass-media turned off "
>
> And I want to have it turned on.
>
> --
> Artur Cichocki
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From kheimerl at cs.berkeley.edu  Mon Jul  4 03:49:35 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sun, 3 Jul 2011 16:49:35 -0700
Subject: [Freeswitch-dev] Chat Dialplan
Message-ID: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>

Hello Freeswitch-dev!

I've recently begin modifying Sofia to support a chat "dialplan", very
similar to the voice dialplan. Basically, a chat/sms will come in at a
certain number/address and be routed via the dialplan. My current plan
for doing this is as follows:

1) Add message events to sofia: I want sofia (and eventually all chat
clients) to send message events through FS's event framework when an
event arrives.
2) Remove message routing from Sofia: Sofia should not be routing
messages itself. Instead, it should just generate the event saying a
message has arrived and someone else should do the routing (namely my
dialplan)
3) Implement a simple chat dialplan module that listens for message
events and routes them according to an XML dialplan.

I have some questions for the FS developer community.

First, building my own module is conceptually simple. Modifying Sofia
is not. What's the best way to go about making these changes? I could
submit extension tickets (e.g., a configuration variable that adds
messages and one that removes routing), or submit patches, or both. I
think this could be broadly useful, and I want to try to contribute to
the community.

Secondly,  I'm having a little difficulty finding the place where
messages are routed in sofia. That's probably the place to insert both
1 & 2. Any pointers would be appreciated.

Thanks!


From aroumie at yahoo.com  Tue Jul  5 11:53:18 2011
From: aroumie at yahoo.com (Ali R.)
Date: Tue, 5 Jul 2011 00:53:18 -0700 (PDT)
Subject: [Freeswitch-dev] mod_socket
In-Reply-To: <853669.65043.qm@web120612.mail.ne1.yahoo.com>
References: <293813.9658.qm@web120604.mail.ne1.yahoo.com>
	<AANLkTik4SNfLA8FZjRE-_SzAkrLuk0dDz6UTft=1Q6EF@mail.gmail.com>
	<750883.12282.qm@web120615.mail.ne1.yahoo.com>
	<30510.52725.qm@web120616.mail.ne1.yahoo.com>
	<AANLkTimC3usNyXZ0_jWOfttSFH0_CNXzOjRWXPU6X1Mp@mail.gmail.com>
	<853669.65043.qm@web120612.mail.ne1.yahoo.com>
Message-ID: <1309852398.40348.YahooMailRC@web120316.mail.ne1.yahoo.com>

Hello All,

Despite my best efforts, I am making no progress in getting channel 
authentication to work through mod_xml_curl. I would appreciate it so much if 
the community could help/guide me on this issue.? Here is a brief summary of the 
scenarios I tried with no success.

Scenario 1:
Created an ACL (acl1) to allow (0.0.0.0/0) with (apply-inbound-acl=acl1 and 
apply-register-acl=acl1) in the internal profile where I thought when an ACL 
authenticated channel hits the dialplan, mod_socket will give me a chance to 
re-authenticate based on the source IP where if the source IP is allowed, I 
continue on.? If the IP is not allowed, I would respond with ?respond (407)? 

Now looking at fs_cli log and Wireshark monitoring, I noticed when I return 
?Respond? with 407 through socket to FS, mod_xml_curl? does not post back any 
directory request to my CGI to re-authenticate based on user/password and 
ultimately the new channel is destroyed.

Scenario 2:
I removed the ACL related settings (mentioned in scenario 1)?I was able to 
achieve my goal but not 100%. For every single REGISTER / INVITE request, 
mod_xml_curl was happily posting to my CGI a directory request where I responded 
with the 

corresponding password for the user and in case my biz logic finds an authorized 
IP, I simply return empty string for the password.? It worked beautifully.? 
However, when FS gets an INVITE request from a UA that is NOT registered 
already, mod_xml_curl does not post anything to my CGI and ultimately the sofia 
respond with ?Proxy Authentication Required? and the channel is dead.
It seems like I needed to create 2 SIP profiles one for ACL authentication and 
another for user/password challenge so mod_xml_curl will post everything to my 
CGI.? I did not try this scenario due to the fact that this will not help me 
since adding new profile requires a new port to listen on which is something 
that breaks my project main requirements where I must keep SIP listening on 
standard port.

Your suggestion and guidance is really appreciated I had already done my 
homework with fs wiki, the book, and the dev-list but no luck?.

Thanks
Ali R





----- Original Message ----
From: Ali R. <aroumie at yahoo.com>
To: freeswitch-dev at lists.freeswitch.org
Sent: Tue, March 29, 2011 9:44:03 AM
Subject: Re: [Freeswitch-dev] mod_socket

Thanks for taking the time to clarify it.? A dedicated web server would scale 
better but this will be a big dependency. I will write a multi-threading module 
just to respond to xml_curl to emulate a web server and see how things perform 
with stress testing....
Thanks Again.




----- Original Message ----
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-dev at lists.freeswitch.org
Sent: Tue, March 29, 2011 7:50:54 AM
Subject: Re: [Freeswitch-dev] mod_socket

You cannot manipulate the user directory over the socket since the
lookup itself is triggered by xml lookups when the phone registers, so
you would need to use xml_curl for this.

You should probably just make a CGI and use a real web server to handle it.



On Mon, Mar 28, 2011 at 11:43 PM, Ali R. <aroumie at yahoo.com> wrote:
> Your suggestion coupled with chapter 10 in the book -page 258-260, I will be
> able to achieve 100% of my goal .? With the "respond" app I can force the
> auth/challenge to take place based on biz logic (that's 50% of my problem 
>fixed)
> and mod_xml_curl? will hand the [user] back to me on a different socket in the
> form of HTTP request where I respond with the password from my biz logic. Now 
>my
> question is, is there a way to negotiate the user/password through mod_socket.
> I'm avoiding to have my app listening for inbound connection to respond to
> curl?.? While at it, FreeSwitch's slogan should be "FreeSwitch understands
> Security" I'm saying so because on page 260 of the FreeSwitch book, I found 
out
> that there is a way that you can pass back the password hashed instead of 
plain
> text which allow me to store it hashed in the database...small thing but 
really
> smart
>
> Regards,
>
>
>
> ----- Original Message ----
> From: Ali R. <aroumie at yahoo.com>
> To: freeswitch-dev at lists.freeswitch.org
> Sent: Mon, March 28, 2011 3:07:42 PM
> Subject: Re: [Freeswitch-dev] mod_socket
>
> Many thanks Anthony for the quick response; You are the hero.
> I will investigate and hope all goes well....
>
>
>
> ----- Original Message ----
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-dev at lists.freeswitch.org
> Sent: Mon, March 28, 2011 2:57:45 PM
> Subject: Re: [Freeswitch-dev] mod_socket
>
> execute the "respond" application with "407" as the arg
>
>
>
>
> On Mon, Mar 28, 2011 at 4:45 PM, Ali R. <aroumie at yahoo.com> wrote:
>> Hi everyone
>> I think my issue could be fixed with mod_xml_curl but I'm using FS purely
>> through sockets (inbound mode)
>> When I get the park event, I execute some biz logic.? Leg's A source IP plays
> a
>> role in this logic so based on the value of the IP (a big pool of allowed IP
>> addresses is generated dynamically and changes very fast so I have a logic to
>> query this pool) all good so far. However, if the incoming channel?s source 
IP
>> is not allowed I would still want to answer and continue on but I must
>>challenge
>> it with a user name and password that I have a logic to retrieve.? Any
> thoughts
>> on how should I go about doing the username/password challenge through the
>> socket without starting a new listening socket for mod_xml_curl requests .? I
>> really appreciate any thoughts on this issue
>>
>> Many Thanks,
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org



? ? ? 

_______________________________________________
FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org



From tculjaga at gmail.com  Tue Jul  5 13:52:27 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 5 Jul 2011 11:52:27 +0200
Subject: [Freeswitch-dev] How to Submit patches?
Message-ID: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>

is there a procedure to do that ?
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From steveayre at gmail.com  Tue Jul  5 13:56:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Jul 2011 10:56:44 +0100
Subject: [Freeswitch-dev] How to Submit patches?
In-Reply-To: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>
References: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>
Message-ID: <CAFiqYumeHoinFBMgHTtDYT7-Wy-z+mZuw0C8BZe3YRjVVkVnKQ@mail.gmail.com>

Yes - file a "New Feature" or "Improvement" bug report at
http://jira.freeswitch.org/ and attach your patch.

It can then be discussed and tracked on Jira, and gets a assigned an ID
which can be used in the commit log when it's added to trunk.

-Steve



On 5 July 2011 10:52, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> is there a procedure to do that ?
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Tue Jul  5 14:02:28 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 5 Jul 2011 12:02:28 +0200
Subject: [Freeswitch-dev] How to Submit patches?
In-Reply-To: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>
References: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E55E00D4@cooper>

Just post them on Jira (http://jira.freeswitch.org/), as soon as they are reviewed and considered ok, they will be commited to git head.

/Peter


Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Tihomir Culjaga
Skickat: den 5 juli 2011 11:52
Till: freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] How to Submit patches?

is there a procedure to do that ?
!DSPAM:4e12dec332761098221006!
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From dome at tel.co.th  Tue Jul  5 14:39:33 2011
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 5 Jul 2011 17:39:33 +0700
Subject: [Freeswitch-dev] api_hangup_hook and channel variable
Message-ID: <CAMFECSAxxu51JE2nNYPTThgn=L-AYHTqEqXC6zWSAHfjNbjSRQ@mail.gmail.com>

This is my lua script

            session:setVariable("session_in_hangup_hook", "true");
            session:setVariable("api_hangup_hook", "lua
hangup/report.lua "..macct);
            session:transfer("dialnow_wh", "XML", "wh");

It's work fine before hangup i cal see session vation in report.lua
but my quession is how to manipulate or create new variable in
report.lua  ?
i want to see variable in cdr

BG

Dome C.


From fdelawarde at wirelessmundi.com  Tue Jul  5 17:42:29 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Jul 2011 15:42:29 +0200
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
Message-ID: <1309873349.17951.255.camel@luna.madrid.commsmundi.com>

Looks nice, count me in for testing once you have something!

Fran?ois.

On Sun, 2011-07-03 at 16:49 -0700, Kurtis Heimerl wrote: 
> Hello Freeswitch-dev!
> 
> I've recently begin modifying Sofia to support a chat "dialplan", very
> similar to the voice dialplan. Basically, a chat/sms will come in at a
> certain number/address and be routed via the dialplan. My current plan
> for doing this is as follows:
> 
> 1) Add message events to sofia: I want sofia (and eventually all chat
> clients) to send message events through FS's event framework when an
> event arrives.
> 2) Remove message routing from Sofia: Sofia should not be routing
> messages itself. Instead, it should just generate the event saying a
> message has arrived and someone else should do the routing (namely my
> dialplan)
> 3) Implement a simple chat dialplan module that listens for message
> events and routes them according to an XML dialplan.
> 
> I have some questions for the FS developer community.
> 
> First, building my own module is conceptually simple. Modifying Sofia
> is not. What's the best way to go about making these changes? I could
> submit extension tickets (e.g., a configuration variable that adds
> messages and one that removes routing), or submit patches, or both. I
> think this could be broadly useful, and I want to try to contribute to
> the community.
> 
> Secondly,  I'm having a little difficulty finding the place where
> messages are routed in sofia. That's probably the place to insert both
> 1 & 2. Any pointers would be appreciated.
> 
> Thanks!
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org




From steveayre at gmail.com  Tue Jul  5 17:48:29 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Jul 2011 14:48:29 +0100
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
Message-ID: <CAFiqYu=iwOzSmUGwc=g7mDVcGFTRQCpC3aNPeSqMofcA6WBn7A@mail.gmail.com>

Have you thought of making it a patch to add a new API interface so that
modules can register chat dialplans, similar to how mod_dialplan_xml works?

-Steve



On 4 July 2011 00:49, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:

> Hello Freeswitch-dev!
>
> I've recently begin modifying Sofia to support a chat "dialplan", very
> similar to the voice dialplan. Basically, a chat/sms will come in at a
> certain number/address and be routed via the dialplan. My current plan
> for doing this is as follows:
>
> 1) Add message events to sofia: I want sofia (and eventually all chat
> clients) to send message events through FS's event framework when an
> event arrives.
> 2) Remove message routing from Soa: Sofia should not be routing
> messages itself. Instead, it should just generate the event saying a
> message has arrived and someone else should do the routing (namely my
> dialplan)
> 3) Implement a simple chat dialplan module that listens for message
> events and routes them according to an XML dialplan.
>
> I have some questions for the FS developer community.
>
> First, building my own module is conceptually simple. Modifying Sofia
> is not. What's the best way to go about making these changes? I could
> submit extension tickets (e.g., a configuration variable that adds
> messages and one that removes routing), or submit patches, or both. I
> think this could be broadly useful, and I want to try to contribute to
> the community.
>
> Secondly,  I'm having a little difficulty finding the place where
> messages are routed in sofia. That's probably the place to insert both
> 1 & 2. Any pointers would be appreciated.
>
> Thanks!
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Jul  5 19:22:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jul 2011 08:22:10 -0700
Subject: [Freeswitch-dev] api_hangup_hook and channel variable
In-Reply-To: <CAMFECSAxxu51JE2nNYPTThgn=L-AYHTqEqXC6zWSAHfjNbjSRQ@mail.gmail.com>
References: <CAMFECSAxxu51JE2nNYPTThgn=L-AYHTqEqXC6zWSAHfjNbjSRQ@mail.gmail.com>
Message-ID: <CAKzWOxWaMG7sR4oMzehLBPEd6Lo38UtB8r9iu2DyLyRHajbc7w@mail.gmail.com>

Once you are in report.lua the channel is already hung up. To the best of my
knowledge there is no way to add/manipulate channel variables in a hangup
hook since you really just have a static copy of the channel variables.

What is it that you are trying to do?

-MC

On Tue, Jul 5, 2011 at 3:39 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> This is my lua script
>
>            session:setVariable("session_in_hangup_hook", "true");
>            session:setVariable("api_hangup_hook", "lua
> hangup/report.lua "..macct);
>            session:transfer("dialnow_wh", "XML", "wh");
>
> It's work fine before hangup i cal see session vation in report.lua
> but my quession is how to manipulate or create new variable in
> report.lua  ?
> i want to see variable in cdr
>
> BG
>
> Dome C.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From rentmycoder at gmail.com  Tue Jul  5 20:31:15 2011
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Tue, 5 Jul 2011 18:31:15 +0200
Subject: [Freeswitch-dev] mod_rtmp
Message-ID: <CAEW218+DngucT+Z-9GJTBDoZA-E2KVqsUod+p49rOqjbKr9zzw@mail.gmail.com>

Hi guys,

any progress on mod_rtmp development???
I'm very excited to start working...

http://jira.freeswitch.org/browse/FS-3368

Wat are we waiting for???

Thanks,
John


From anthony.minessale at gmail.com  Tue Jul  5 20:56:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jul 2011 11:56:38 -0500
Subject: [Freeswitch-dev] mod_rtmp
In-Reply-To: <CAEW218+DngucT+Z-9GJTBDoZA-E2KVqsUod+p49rOqjbKr9zzw@mail.gmail.com>
References: <CAEW218+DngucT+Z-9GJTBDoZA-E2KVqsUod+p49rOqjbKr9zzw@mail.gmail.com>
Message-ID: <CAKbxfG_7m4vPH0KCWVdF0ho2UQNm699DWVhdVCowSWb85Deeew@mail.gmail.com>

In fact I checked in the code this morning that demonstrates the basic
functionality.
I think this should be used as a reference and rewritten.


On Tue, Jul 5, 2011 at 11:31 AM, rentmycoder rentmycoder
<rentmycoder at gmail.com> wrote:
> Hi guys,
>
> any progress on mod_rtmp development???
> I'm very excited to start working...
>
> http://jira.freeswitch.org/browse/FS-3368
>
> Wat are we waiting for???
>
> Thanks,
> John
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From zhongxiang721 at gmail.com  Sun Jul  3 05:12:44 2011
From: zhongxiang721 at gmail.com (salzh)
Date: Sun, 3 Jul 2011 09:12:44 +0800
Subject: [Freeswitch-dev] get answertime dialstatus after bridge
In-Reply-To: <CAM3SFgDg+rinmVT7-geLxNPxvVGzNF=CzgfafUuGfB+OGPEp=w@mail.gmail.com>
References: <CAM3SFgDg+rinmVT7-geLxNPxvVGzNF=CzgfafUuGfB+OGPEp=w@mail.gmail.com>
Message-ID: <CAM3SFgBVFogDGdwOvJpFjsngZO3s4w40rWkdVZObyNhrT56zXQ@mail.gmail.com>

---------- Forwarded message ----------
From: salzh <zhongxiang721 at gmail.com>
Date: 2011/7/3
Subject: get answertime dialstatus after bridge
To: freeswitch-users at lists.freeswitch.org


hi.
is there any way to get a call information ?such as answertime
dialstatus and so on after executing 'bridge' application? just like
what the application 'dial' do in asterisk.


From alhakeem at gmail.com  Mon Jul  4 12:48:31 2011
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Mon, 4 Jul 2011 09:48:31 +0100
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAK9eRpOA7gpCrwOkcz9SJx8BAAAAAA==@gmail.com>

Kurtis,
Which protocol will this chat/sms dialplan be compliant with ?.
Sip MESSAGE method , SIMPLE and XMPP are both widely adopted.
It would be great to add support for connectivity to external domains such as
Gtalk, Yahoo Messenger etc.
Cheers,
AH

-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Kurtis Heimerl
Sent: Monday, July 04, 2011 12:50 AM
To: freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] Chat Dialplan

Hello Freeswitch-dev!

I've recently begin modifying Sofia to support a chat "dialplan", very
similar to the voice dialplan. Basically, a chat/sms will come in at a
certain number/address and be routed via the dialplan. My current plan
for doing this is as follows:

1) Add message events to sofia: I want sofia (and eventually all chat
clients) to send message events through FS's event framework when an
event arrives.
2) Remove message routing from Sofia: Sofia should not be routing
messages itself. Instead, it should just generate the event saying a
message has arrived and someone else should do the routing (namely my
dialplan)
3) Implement a simple chat dialplan module that listens for message
events and routes them according to an XML dialplan.

I have some questions for the FS developer community.

First, building my own module is conceptually simple. Modifying Sofia
is not. What's the best way to go about making these changes? I could
submit extension tickets (e.g., a configuration variable that adds
messages and one that removes routing), or submit patches, or both. I
think this could be broadly useful, and I want to try to contribute to
the community.

Secondly,  I'm having a little difficulty finding the place where
messages are routed in sofia. That's probably the place to insert both
1 & 2. Any pointers would be appreciated.

Thanks!

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org




From leucaruth at gmail.com  Mon Jul  4 15:48:36 2011
From: leucaruth at gmail.com (Nacho)
Date: Mon, 4 Jul 2011 13:48:36 +0200
Subject: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
Message-ID: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>

Hello all.

My name is Jose. This is my first post here. I joined the list because I was
wondering about developing a feature for Freeswitch and I would like to know
your opinion.

The idea is about making a non-persistent client for cell phones that works
with PUSH technology (C2DM and Android for example). These clients wouldn't
be connected to the Freeswitch server at first, but if there is an incoming
call, the Freeswitch server would send a PUSH message to these clients, the
client would process it and, if accepted by the user, the client would
awake, connect to Freeswitch and then receive the the call invite message
and accept it automatically.

I don't know if this idea is plausible due to the real time restrictions we
have to face in phone calls. If the delay introduced by C2DM delivery is
high, the waiting time for the caller is probably something that he isn't
willing to accept in order to get his call answered.

Another question is about the battery life time it would save because the
application woulnd't need to be answering the "still alive" ACKS from
Freeswitch.

I'm new in freeswitch developing and still learning about it, but I'm really
interested in this, so any help would be kindly appreciated.

-- 
Aquellos que hablan son esclavos de sus palabras y los que callan due?os de
su silencio.
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From acichocki at supermedia.pl  Mon Jul  4 21:41:38 2011
From: acichocki at supermedia.pl (Artur Cichocki)
Date: Mon, 04 Jul 2011 19:41:38 +0200
Subject: [Freeswitch-dev] Removing SDP from 183/180
In-Reply-To: <CAPHrvXFY1rLUeicg_Cn7Q2eMFNLPGds1uNejxaknCF+gcqvUTQ@mail.gmail.com>
References: <4DDF5CEE.5060505@supermedia.pl>	<BANLkTik2OqWE+4BPRD08fBXrnCxwY1cfZg@mail.gmail.com>	<4E0B1566.8000903@supermedia.pl>
	<CAPHrvXFY1rLUeicg_Cn7Q2eMFNLPGds1uNejxaknCF+gcqvUTQ@mail.gmail.com>
Message-ID: <4E11FB52.3060305@supermedia.pl>

W dniu 02.07.2011 13:16, Prashant Lamba pisze:
>      * Be certain that you have inbound-bypass-media turned off "
> And I want to have it turned on.
>
> If you need to not send 183, then you MUST have bypass_media enabled. I
> have tested it and it works. Just so you know, if FS is not doing media,
> that means its not sending 183, hence someone (other app) needs to do
> it. Hence bypass_media need to be true
That's not the point. Check my original mail please.

The patch is to deny media before real connection establishment with 
bypas_media on. And that functionality doesn't exist in current FS code.

-- 
Artur Cichocki


From anthony.minessale at gmail.com  Tue Jul  5 22:18:59 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jul 2011 13:18:59 -0500
Subject: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
In-Reply-To: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>
References: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>
Message-ID: <CAKbxfG_7D+qE6wY-ob=QNMaiZn=WxhAXGAkocU9dNgqHtYHxWA@mail.gmail.com>

it would certainly be worth trying, you can use generated indications
to give the caller ringing or other feedback while you do it.


On Mon, Jul 4, 2011 at 6:48 AM, Nacho <leucaruth at gmail.com> wrote:
> Hello all.
> My name is Jose. This is my first post here. I joined the list because I was
> wondering about developing a feature for Freeswitch and I would like to know
> your opinion.
> The idea is about making a non-persistent client for cell phones that works
> with PUSH technology (C2DM and Android for example). These clients wouldn't
> be connected to the Freeswitch server at first, but if there is an incoming
> call, the Freeswitch server would send a PUSH message to these clients, the
> client would process it and, if accepted by the user, the client would
> awake, connect to Freeswitch and then receive the the call invite message
> and accept it automatically.
> I don't know if this idea is plausible due to the real time restrictions we
> have to face in phone calls. If the delay introduced by C2DM delivery is
> high, the waiting time for the caller is probably something that he isn't
> willing to accept in order to get his call answered.
> Another question is about the battery life time it would save because the
> application woulnd't need to be answering the "still alive" ACKS from
> Freeswitch.
> I'm new in freeswitch developing and still learning about it, but I'm really
> interested in this, so any help would be kindly appreciated.
> --
> Aquellos que hablan son esclavos de sus palabras y los que callan due?os de
> su silencio.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From kris at kriskinc.com  Tue Jul  5 22:20:58 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 5 Jul 2011 14:20:58 -0400
Subject: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
In-Reply-To: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>
References: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>
Message-ID: <CAKDfjgce=KSUY84+c5LwnqwSR2OrRVSwyM+BgKX=JcqVzK4JpA@mail.gmail.com>

Jose,

  Whether it's C2DM or APNS (Apple) this functionality is best left
out of FreeSWITCH.  FreeSWITCH has plenty of other existing means
(XML_CURL, socket, etc) to drive dynamic call functionality and
interact with other technologies.

  How do you ask?  An imcoming call to FreeSWITCH could hit static
dialplan and execute a socket connection to some other custom server.
This server could tell FreeSWITCH to do something (play media,
ringback, messages, whatever) while it sends a PUSH (via C2DM or APNS,
for example).

  One could implement such a socket program easily.  It can be
extended and maintained separately as new push technologies become
available.

On Mon, Jul 4, 2011 at 7:48 AM, Nacho <leucaruth at gmail.com> wrote:
> Hello all.
> My name is Jose. This is my first post here. I joined the list because I was
> wondering about developing a feature for Freeswitch and I would like to know
> your opinion.
> The idea is about making a non-persistent client for cell phones that works
> with PUSH technology (C2DM and Android for example). These clients wouldn't
> be connected to the Freeswitch server at first, but if there is an incoming
> call, the Freeswitch server would send a PUSH message to these clients, the
> client would process it and, if accepted by the user, the client would
> awake, connect to Freeswitch and then receive the the call invite message
> and accept it automatically.
> I don't know if this idea is plausible due to the real time restrictions we
> have to face in phone calls. If the delay introduced by C2DM delivery is
> high, the waiting time for the caller is probably something that he isn't
> willing to accept in order to get his call answered.
> Another question is about the battery life time it would save because the
> application woulnd't need to be answering the "still alive" ACKS from
> Freeswitch.
> I'm new in freeswitch developing and still learning about it, but I'm really
> interested in this, so any help would be kindly appreciated.
> --
> Aquellos que hablan son esclavos de sus palabras y los que callan due?os de
> su silencio.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From steveayre at gmail.com  Tue Jul  5 22:21:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Jul 2011 19:21:23 +0100
Subject: [Freeswitch-dev] Removing SDP from 183/180
In-Reply-To: <4E0B1566.8000903@supermedia.pl>
References: <4DDF5CEE.5060505@supermedia.pl>
	<BANLkTik2OqWE+4BPRD08fBXrnCxwY1cfZg@mail.gmail.com>
	<4E0B1566.8000903@supermedia.pl>
Message-ID: <CAFiqYumqg5vZYQbdAnS8qbRKyYmWg1Rbmhqp-1mOxEfEY2nJPw@mail.gmail.com>

>
> And I want to have it turned on.
>

Be aware then that if FS were to ignore the 180 and not pass it along to the
caller, then the callee would already be sending the early media directly to
the caller. They'll ignore it sure but it'll still be consuming their
bandwidth and be visible to the user if they wanted to use something like
Wireshark.


-Steve



On 29 June 2011 13:07, Artur Cichocki <acichocki at supermedia.pl> wrote:

> W dniu 27.06.2011 21:58, Anthony Minessale pisze:
> > this functionality is already implemented with
> {ignore_early_media=ring_ready}
> >
> I tried all combinations of ignore_early_media with bybass-media, it
> didn't work.
>
> According to http://wiki.freeswitch.org/wiki/Early_Media:
> " Troubleshooting
>
>     * Be certain that you have inbound-bypass-media turned off "
>
> And I want to have it turned on.
>
> --
> Artur Cichocki
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From kris at kriskinc.com  Tue Jul  5 22:23:05 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 5 Jul 2011 14:23:05 -0400
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAK9eRpOA7gpCrwOkcz9SJx8BAAAAAA==@gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAK9eRpOA7gpCrwOkcz9SJx8BAAAAAA==@gmail.com>
Message-ID: <CAKDfjgeLLcJVQjkDrU2dS9hFn-4xCYQ2gzDmxXnT0XSf4aO6+Q@mail.gmail.com>

To be clear SIMPLE is a collection of standards that includes message
exchange using the MESSAGE method.

On Mon, Jul 4, 2011 at 4:48 AM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> Kurtis,
> Which protocol will this chat/sms dialplan be compliant with ?.
> Sip MESSAGE method , SIMPLE and XMPP are both widely adopted.
> It would be great to add support for connectivity to external domains such as
> Gtalk, Yahoo Messenger etc.
> Cheers,
> AH

-- 
Kristian Kielhofner


From anthony.minessale at gmail.com  Tue Jul  5 23:07:45 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jul 2011 14:07:45 -0500
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
Message-ID: <CAKbxfG8jaY61V5w2SH7BJ1zgqD49FfKwC32m7-yqK2pvnCD7Qw@mail.gmail.com>

This sounds redundant and intrusive.
We already have a concept of namespace within the chat username which
is used for routing.

we prefix the namespace into the user id such as conf+1234 at domain.com
This allows conference to get messages from sip, xmpp or any other
protocol that supports chat.

When a from user on chat contains +:

each module just parses the inbound chat message, cuts off the proto
"everything up to the first +" in this case conf
and routes it to the module registered to that namespace.

Right now conference will advertise it's presence and receive chat of
the word list and respond with the list despite the protocol used, eg
sip, jabber etc.

Also chat messages are routed to individual calls when they are
associated with an active channel allowing you to use chat inside IVR.

I recommend trying to understand all of this before proposing drastic
changes that will regress all of this functionality.






On Sun, Jul 3, 2011 at 6:49 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
> Hello Freeswitch-dev!
>
> I've recently begin modifying Sofia to support a chat "dialplan", very
> similar to the voice dialplan. Basically, a chat/sms will come in at a
> certain number/address and be routed via the dialplan. My current plan
> for doing this is as follows:
>
> 1) Add message events to sofia: I want sofia (and eventually all chat
> clients) to send message events through FS's event framework when an
> event arrives.
> 2) Remove message routing from Sofia: Sofia should not be routing
> messages itself. Instead, it should just generate the event saying a
> message has arrived and someone else should do the routing (namely my
> dialplan)
> 3) Implement a simple chat dialplan module that listens for message
> events and routes them according to an XML dialplan.
>
> I have some questions for the FS developer community.
>
> First, building my own module is conceptually simple. Modifying Sofia
> is not. What's the best way to go about making these changes? I could
> submit extension tickets (e.g., a configuration variable that adds
> messages and one that removes routing), or submit patches, or both. I
> think this could be broadly useful, and I want to try to contribute to
> the community.
>
> Secondly, ?I'm having a little difficulty finding the place where
> messages are routed in sofia. That's probably the place to insert both
> 1 & 2. Any pointers would be appreciated.
>
> Thanks!
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From tculjaga at gmail.com  Wed Jul  6 01:32:31 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 5 Jul 2011 23:32:31 +0200
Subject: [Freeswitch-dev] How to Submit patches?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E55E00D4@cooper>
References: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E55E00D4@cooper>
Message-ID: <CAJoKKdXgbHt56c5BW8duRLyPATyHik4fjN02E6U5GoMExoKxhw@mail.gmail.com>

thanks for the answers ... really appreciated.

yes its an improvement on mod_sofia to handle diversion headers within 3xx
messages and a patch on sofia stack to allow multiple Diversion headers...
right now, if there are more than one.. sofia will use the last Diversion
found in a sip message...


will post it on jira so you can comment it (reject or accept)

many thanks,
T.



On Tue, Jul 5, 2011 at 12:02 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Just post them on Jira (http://jira.freeswitch.org/), as soon as they are
> reviewed and considered ok, they will be commited to git head.****
>
> ** **
>
> /Peter****
>
> ** **
>
> ** **
>
> *Fr?n:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *F?r *Tihomir Culjaga
> *Skickat:* den 5 juli 2011 11:52
> *Till:* freeswitch-dev at lists.freeswitch.org
> *?mne:* [Freeswitch-dev] How to Submit patches?****
>
> ** **
>
> is there a procedure to do that ?
> !DSPAM:4e12dec332761098221006! ****
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Jul  6 01:49:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jul 2011 16:49:26 -0500
Subject: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
In-Reply-To: <CAKDfjgce=KSUY84+c5LwnqwSR2OrRVSwyM+BgKX=JcqVzK4JpA@mail.gmail.com>
References: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>
	<CAKDfjgce=KSUY84+c5LwnqwSR2OrRVSwyM+BgKX=JcqVzK4JpA@mail.gmail.com>
Message-ID: <CAKbxfG_hNNWf_jC5pT_-9uUpALjuZTBNJE2saSG_K-4-HFQY0g@mail.gmail.com>

it can be left out of FS code but still part of contrib if an external
solution is developed.


On Tue, Jul 5, 2011 at 1:20 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> Jose,
>
> ?Whether it's C2DM or APNS (Apple) this functionality is best left
> out of FreeSWITCH. ?FreeSWITCH has plenty of other existing means
> (XML_CURL, socket, etc) to drive dynamic call functionality and
> interact with other technologies.
>
> ?How do you ask? ?An imcoming call to FreeSWITCH could hit static
> dialplan and execute a socket connection to some other custom server.
> This server could tell FreeSWITCH to do something (play media,
> ringback, messages, whatever) while it sends a PUSH (via C2DM or APNS,
> for example).
>
> ?One could implement such a socket program easily. ?It can be
> extended and maintained separately as new push technologies become
> available.
>
> On Mon, Jul 4, 2011 at 7:48 AM, Nacho <leucaruth at gmail.com> wrote:
>> Hello all.
>> My name is Jose. This is my first post here. I joined the list because I was
>> wondering about developing a feature for Freeswitch and I would like to know
>> your opinion.
>> The idea is about making a non-persistent client for cell phones that works
>> with PUSH technology (C2DM and Android for example). These clients wouldn't
>> be connected to the Freeswitch server at first, but if there is an incoming
>> call, the Freeswitch server would send a PUSH message to these clients, the
>> client would process it and, if accepted by the user, the client would
>> awake, connect to Freeswitch and then receive the the call invite message
>> and accept it automatically.
>> I don't know if this idea is plausible due to the real time restrictions we
>> have to face in phone calls. If the delay introduced by C2DM delivery is
>> high, the waiting time for the caller is probably something that he isn't
>> willing to accept in order to get his call answered.
>> Another question is about the battery life time it would save because the
>> application woulnd't need to be answering the "still alive" ACKS from
>> Freeswitch.
>> I'm new in freeswitch developing and still learning about it, but I'm really
>> interested in this, so any help would be kindly appreciated.
>> --
>> Aquellos que hablan son esclavos de sus palabras y los que callan due?os de
>> su silencio.
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From rhuddleston at gmail.com  Wed Jul  6 02:26:37 2011
From: rhuddleston at gmail.com (Robert-iPhone)
Date: Tue, 5 Jul 2011 18:26:37 -0400
Subject: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
In-Reply-To: <CAKbxfG_hNNWf_jC5pT_-9uUpALjuZTBNJE2saSG_K-4-HFQY0g@mail.gmail.com>
References: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>
	<CAKDfjgce=KSUY84+c5LwnqwSR2OrRVSwyM+BgKX=JcqVzK4JpA@mail.gmail.com>
	<CAKbxfG_hNNWf_jC5pT_-9uUpALjuZTBNJE2saSG_K-4-HFQY0g@mail.gmail.com>
Message-ID: <35195A50-B87B-4970-BA7E-359849EDE8B2@gmail.com>

I have interest too! Considering mwi / voicemail push for apple

Sent from my iPhone

On Jul 5, 2011, at 5:49 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> it can be left out of FS code but still part of contrib if an external
> solution is developed.
> 
> 
> On Tue, Jul 5, 2011 at 1:20 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>> Jose,
>> 
>>  Whether it's C2DM or APNS (Apple) this functionality is best left
>> out of FreeSWITCH.  FreeSWITCH has plenty of other existing means
>> (XML_CURL, socket, etc) to drive dynamic call functionality and
>> interact with other technologies.
>> 
>>  How do you ask?  An imcoming call to FreeSWITCH could hit static
>> dialplan and execute a socket connection to some other custom server.
>> This server could tell FreeSWITCH to do something (play media,
>> ringback, messages, whatever) while it sends a PUSH (via C2DM or APNS,
>> for example).
>> 
>>  One could implement such a socket program easily.  It can be
>> extended and maintained separately as new push technologies become
>> available.
>> 
>> On Mon, Jul 4, 2011 at 7:48 AM, Nacho <leucaruth at gmail.com> wrote:
>>> Hello all.
>>> My name is Jose. This is my first post here. I joined the list because I was
>>> wondering about developing a feature for Freeswitch and I would like to know
>>> your opinion.
>>> The idea is about making a non-persistent client for cell phones that works
>>> with PUSH technology (C2DM and Android for example). These clients wouldn't
>>> be connected to the Freeswitch server at first, but if there is an incoming
>>> call, the Freeswitch server would send a PUSH message to these clients, the
>>> client would process it and, if accepted by the user, the client would
>>> awake, connect to Freeswitch and then receive the the call invite message
>>> and accept it automatically.
>>> I don't know if this idea is plausible due to the real time restrictions we
>>> have to face in phone calls. If the delay introduced by C2DM delivery is
>>> high, the waiting time for the caller is probably something that he isn't
>>> willing to accept in order to get his call answered.
>>> Another question is about the battery life time it would save because the
>>> application woulnd't need to be answering the "still alive" ACKS from
>>> Freeswitch.
>>> I'm new in freeswitch developing and still learning about it, but I'm really
>>> interested in this, so any help would be kindly appreciated.
>>> --
>>> Aquellos que hablan son esclavos de sus palabras y los que callan due?os de
>>> su silencio.
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> 
>> 
>> --
>> Kristian Kielhofner
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org


From jaybinks at gmail.com  Wed Jul  6 03:49:48 2011
From: jaybinks at gmail.com (jay binks)
Date: Wed, 6 Jul 2011 09:49:48 +1000
Subject: [Freeswitch-dev] How to Submit patches?
In-Reply-To: <CAJoKKdXgbHt56c5BW8duRLyPATyHik4fjN02E6U5GoMExoKxhw@mail.gmail.com>
References: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E55E00D4@cooper>
	<CAJoKKdXgbHt56c5BW8duRLyPATyHik4fjN02E6U5GoMExoKxhw@mail.gmail.com>
Message-ID: <CAO47Hpegj70-Ep-vPDs6SCPFZ7M2VZLUhivC3NfyQMu1eQJNrg@mail.gmail.com>

would be interested to look at this...
what jira ticket is it ?

Jay

On Wed, Jul 6, 2011 at 7:32 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> thanks for the answers ... really appreciated.
>
> yes its an improvement on mod_sofia to handle diversion headers within 3xx
> messages and a patch on sofia stack to allow multiple Diversion headers...
> right now, if there are more than one.. sofia will use the last Diversion
> found in a sip message...
>
>
> will post it on jira so you can comment it (reject or accept)
>
> many thanks,
> T.
>
>
>
> On Tue, Jul 5, 2011 at 12:02 PM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>> Just post them on Jira (http://jira.freeswitch.org/), as soon as they are
>> reviewed and considered ok, they will be commited to git head.****
>>
>> ** **
>>
>> /Peter****
>>
>> ** **
>>
>> ** **
>>
>> *Fr?n:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
>> freeswitch-dev-bounces at lists.freeswitch.org] *F?r *Tihomir Culjaga
>> *Skickat:* den 5 juli 2011 11:52
>> *Till:* freeswitch-dev at lists.freeswitch.org
>> *?mne:* [Freeswitch-dev] How to Submit patches?****
>>
>> ** **
>>
>> is there a procedure to do that ?
>> !DSPAM:4e12dec332761098221006! ****
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From msc at freeswitch.org  Wed Jul  6 03:56:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jul 2011 16:56:30 -0700
Subject: [Freeswitch-dev] How to Submit patches?
In-Reply-To: <CAO47Hpegj70-Ep-vPDs6SCPFZ7M2VZLUhivC3NfyQMu1eQJNrg@mail.gmail.com>
References: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E55E00D4@cooper>
	<CAJoKKdXgbHt56c5BW8duRLyPATyHik4fjN02E6U5GoMExoKxhw@mail.gmail.com>
	<CAO47Hpegj70-Ep-vPDs6SCPFZ7M2VZLUhivC3NfyQMu1eQJNrg@mail.gmail.com>
Message-ID: <CAKzWOxXxcKs=gkS9OwZVa4KO3JEVf5Sjsu-aW80w=3kpLM2KDQ@mail.gmail.com>

On Tue, Jul 5, 2011 at 4:49 PM, jay binks <jaybinks at gmail.com> wrote:

> would be interested to look at this...
> what jira ticket is it ?
>
> It's in the jira at this link:
http://bit.ly/2cK1U

-MC
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From jaybinks at gmail.com  Wed Jul  6 04:25:04 2011
From: jaybinks at gmail.com (jay binks)
Date: Wed, 6 Jul 2011 10:25:04 +1000
Subject: [Freeswitch-dev] How to Submit patches?
In-Reply-To: <CAKzWOxXxcKs=gkS9OwZVa4KO3JEVf5Sjsu-aW80w=3kpLM2KDQ@mail.gmail.com>
References: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E55E00D4@cooper>
	<CAJoKKdXgbHt56c5BW8duRLyPATyHik4fjN02E6U5GoMExoKxhw@mail.gmail.com>
	<CAO47Hpegj70-Ep-vPDs6SCPFZ7M2VZLUhivC3NfyQMu1eQJNrg@mail.gmail.com>
	<CAKzWOxXxcKs=gkS9OwZVa4KO3JEVf5Sjsu-aW80w=3kpLM2KDQ@mail.gmail.com>
Message-ID: <CAO47HpdCpp_X1-RASuXbDrsFWruKuq9zFocVcsONgGKS+6N9RA@mail.gmail.com>

haha .... I like that one  :)

On Wed, Jul 6, 2011 at 9:56 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Jul 5, 2011 at 4:49 PM, jay binks <jaybinks at gmail.com> wrote:
>
>> would be interested to look at this...
>> what jira ticket is it ?
>>
>> It's in the jira at this link:
> http://bit.ly/2cK1U
>
> -MC
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From msc at freeswitch.org  Wed Jul  6 04:33:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jul 2011 17:33:24 -0700
Subject: [Freeswitch-dev] How to Submit patches?
In-Reply-To: <CAO47HpdCpp_X1-RASuXbDrsFWruKuq9zFocVcsONgGKS+6N9RA@mail.gmail.com>
References: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E55E00D4@cooper>
	<CAJoKKdXgbHt56c5BW8duRLyPATyHik4fjN02E6U5GoMExoKxhw@mail.gmail.com>
	<CAO47Hpegj70-Ep-vPDs6SCPFZ7M2VZLUhivC3NfyQMu1eQJNrg@mail.gmail.com>
	<CAKzWOxXxcKs=gkS9OwZVa4KO3JEVf5Sjsu-aW80w=3kpLM2KDQ@mail.gmail.com>
	<CAO47HpdCpp_X1-RASuXbDrsFWruKuq9zFocVcsONgGKS+6N9RA@mail.gmail.com>
Message-ID: <CAKzWOxUYRpWPKa3dFAaz5KU4D5_bxAfC9U1cqQGG3etGzrYnVA@mail.gmail.com>

On Tue, Jul 5, 2011 at 5:25 PM, jay binks <jaybinks at gmail.com> wrote:

> haha .... I like that one  :)
>

That was your personal invitation. ;)
-MC
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From kheimerl at cs.berkeley.edu  Wed Jul  6 06:31:54 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Tue, 5 Jul 2011 19:31:54 -0700
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
Message-ID: <CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>

I forgot to disable digest mode, and so I have 5 emails to respond to
in 2 received mails. Instead, I'm going to address the highlights
here:

Steven Ayre <steveayre at gmail.com> said:
"Have you thought of making it a patch to add a new API interface so
that modules can register chat dialplans, similar to how
mod_dialplan_xml works?

-Steve"

I don't quite understand what you're saying here. Individual modules
(e.g., Sofia) would register a different dialplan? Is that how it
currently works? I thought, if they selected an XML dialplan in the
conf, they all used the same one. I'm obviously confused, so please
explain in more depth!

"Abdul Hakeem" <alhakeem at gmail.com> said:
"Kurtis,
Which protocol will this chat/sms dialplan be compliant with ?.
Sip MESSAGE method , SIMPLE and XMPP are both widely adopted.
It would be great to add support for connectivity to external domains such as
Gtalk, Yahoo Messenger etc.
Cheers,
AH"

So far as my current architectural vision, any module implementing a
MESSAGE event should be able to trigger a dialplan action. My plan is
to start with Sofia, so it would support SIMPLE. Others should be
pretty easy as well.

Lastly,
Anthony Minessale <anthony.minessale at gmail.com> said:
"This sounds redundant and intrusive.
We already have a concept of namespace within the chat username which
is used for routing.

we prefix the namespace into the user id such as conf+1234 at domain.com
This allows conference to get messages from sip, xmpp or any other
protocol that supports chat.

When a from user on chat contains +:

each module just parses the inbound chat message, cuts off the proto
"everything up to the first +" in this case conf
and routes it to the module registered to that namespace.

Right now conference will advertise it's presence and receive chat of
the word list and respond with the list despite the protocol used, eg
sip, jabber etc.

Also chat messages are routed to individual calls when they are
associated with an active channel allowing you to use chat inside IVR.

I recommend trying to understand all of this before proposing drastic
changes that will regress all of this functionality."

I need a programmatic layer for routing messages. What's currently in
FS is (seemingly) unable to do this. I'm aware that the current system
routes messages to users correctly; I don't think it enables any sort
of interesting programmatic access. For instance, building a system
that receives a message and immediately starts a call is not something
I think can be currently done. The dialplan abstraction is
well-understood by FS users, and an extension I think could provide a
lot of value.

I don't think these changes will regress any functionality either. I
proposed adding a configuration flag that switched between behaviors,
so that existing functionality won't be modified. Lastly, sending this
email was part of me trying to understand the current system. Thanks
for the direction.

If I'm wrong, and there's a simple way to provide programmatic access,
I'd love to hear about it. That's a lot less work for me!

On Sun, Jul 3, 2011 at 4:49 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
> Hello Freeswitch-dev!
>
> I've recently begin modifying Sofia to support a chat "dialplan", very
> similar to the voice dialplan. Basically, a chat/sms will come in at a
> certain number/address and be routed via the dialplan. My current plan
> for doing this is as follows:
>
> 1) Add message events to sofia: I want sofia (and eventually all chat
> clients) to send message events through FS's event framework when an
> event arrives.
> 2) Remove message routing from Sofia: Sofia should not be routing
> messages itself. Instead, it should just generate the event saying a
> message has arrived and someone else should do the routing (namely my
> dialplan)
> 3) Implement a simple chat dialplan module that listens for message
> events and routes them according to an XML dialplan.
>
> I have some questions for the FS developer community.
>
> First, building my own module is conceptually simple. Modifying Sofia
> is not. What's the best way to go about making these changes? I could
> submit extension tickets (e.g., a configuration variable that adds
> messages and one that removes routing), or submit patches, or both. I
> think this could be broadly useful, and I want to try to contribute to
> the community.
>
> Secondly, ?I'm having a little difficulty finding the place where
> messages are routed in sofia. That's probably the place to insert both
> 1 & 2. Any pointers would be appreciated.
>
> Thanks!
>


From jmesquita at freeswitch.org  Wed Jul  6 08:59:37 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 6 Jul 2011 01:59:37 -0300
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
Message-ID: <CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>

I believe that these messages show up as events on ESL, don't they?
Therefore you can do whatever from there...

Regards,
Jo?o Mesquita



On Tue, Jul 5, 2011 at 11:31 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>wrote:

> I forgot to disable digest mode, and so I have 5 emails to respond to
> in 2 received mails. Instead, I'm going to address the highlights
> here:
>
> Steven Ayre <steveayre at gmail.com> said:
> "Have you thought of making it a patch to add a new API interface so
> that modules can register chat dialplans, similar to how
> mod_dialplan_xml works?
>
> -Steve"
>
> I don't quite understand what you're saying here. Individual modules
> (e.g., Sofia) would register a different dialplan? Is that how it
> currently works? I thought, if they selected an XML dialplan in the
> conf, they all used the same one. I'm obviously confused, so please
> explain in more depth!
>
> "Abdul Hakeem" <alhakeem at gmail.com> said:
> "Kurtis,
> Which protocol will this chat/sms dialplan be compliant with ?.
> Sip MESSAGE method , SIMPLE and XMPP are both widely adopted.
> It would be great to add support for connectivity to external domains such
> as
> Gtalk, Yahoo Messenger etc.
> Cheers,
> AH"
>
> So far as my current architectural vision, any module implementing a
> MESSAGE event should be able to trigger a dialplan action. My plan is
> to start with Sofia, so it would support SIMPLE. Others should be
> pretty easy as well.
>
> Lastly,
> Anthony Minessale <anthony.minessale at gmail.com> said:
> "This sounds redundant and intrusive.
> We already have a concept of namespace within the chat username which
> is used for routing.
>
> we prefix the namespace into the user id such as conf+1234 at domain.com
> This allows conference to get messages from sip, xmpp or any other
> protocol that supports chat.
>
> When a from user on chat contains +:
>
> each module just parses the inbound chat message, cuts off the proto
> "everything up to the first +" in this case conf
> and routes it to the module registered to that namespace.
>
> Right now conference will advertise it's presence and receive chat of
> the word list and respond with the list despite the protocol used, eg
> sip, jabber etc.
>
> Also chat messages are routed to individual calls when they are
> associated with an active channel allowing you to use chat inside IVR.
>
> I recommend trying to understand all of this before proposing drastic
> changes that will regress all of this functionality."
>
> I need a programmatic layer for routing messages. What's currently in
> FS is (seemingly) unable to do this. I'm aware that the current system
> routes messages to users correctly; I don't think it enables any sort
> of interesting programmatic access. For instance, building a system
> that receives a message and immediately starts a call is not something
> I think can be currently done. The dialplan abstraction is
> well-understood by FS users, and an extension I think could provide a
> lot of value.
>
> I don't think these changes will regress any functionality either. I
> proposed adding a configuration flag that switched between behaviors,
> so that existing functionality won't be modified. Lastly, sending this
> email was part of me trying to understand the current system. Thanks
> for the direction.
>
> If I'm wrong, and there's a simple way to provide programmatic access,
> I'd love to hear about it. That's a lot less work for me!
>
> On Sun, Jul 3, 2011 at 4:49 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
> wrote:
> > Hello Freeswitch-dev!
> >
> > I've recently begin modifying Sofia to support a chat "dialplan", very
> > similar to the voice dialplan. Basically, a chat/sms will come in at a
> > certain number/address and be routed via the dialplan. My current plan
> > for doing this is as follows:
> >
> > 1) Add message events to sofia: I want sofia (and eventually all chat
> > clients) to send message events through FS's event framework when an
> > event arrives.
> > 2) Remove message routing from Sofia: Sofia should not be routing
> > messages itself. Instead, it should just generate the event saying a
> > message has arrived and someone else should do the routing (namely my
> > dialplan)
> > 3) Implement a simple chat dialplan module that listens for message
> > events and routes them according to an XML dialplan.
> >
> > I have some questions for the FS developer community.
> >
> > First, building my own module is conceptually simple. Modifying Sofia
> > is not. What's the best way to go about making these changes? I could
> > submit extension tickets (e.g., a configuration variable that adds
> > messages and one that removes routing), or submit patches, or both. I
> > think this could be broadly useful, and I want to try to contribute to
> > the community.
> >
> > Secondly,  I'm having a little difficulty finding the place where
> > messages are routed in sofia. That's probably the place to insert both
> > 1 & 2. Any pointers would be appreciated.
> >
> > Thanks!
> >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From kheimerl at cs.berkeley.edu  Wed Jul  6 09:20:56 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Tue, 5 Jul 2011 22:20:56 -0700
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
Message-ID: <CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>

I don't think so. I set the ESL to log all events (/event all) from
the command line and received no events when sofia received SIMPLE
messages. Is that a bug? I had assumed it's intentional.

2011/7/5 Jo?o Mesquita <jmesquita at freeswitch.org>:
> I believe that these messages show up as events on ESL, don't they?
> Therefore you can do whatever from there...
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Tue, Jul 5, 2011 at 11:31 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
> wrote:
>>
>> I forgot to disable digest mode, and so I have 5 emails to respond to
>> in 2 received mails. Instead, I'm going to address the highlights
>> here:
>>
>> Steven Ayre <steveayre at gmail.com> said:
>> "Have you thought of making it a patch to add a new API interface so
>> that modules can register chat dialplans, similar to how
>> mod_dialplan_xml works?
>>
>> -Steve"
>>
>> I don't quite understand what you're saying here. Individual modules
>> (e.g., Sofia) would register a different dialplan? Is that how it
>> currently works? I thought, if they selected an XML dialplan in the
>> conf, they all used the same one. I'm obviously confused, so please
>> explain in more depth!
>>
>> "Abdul Hakeem" <alhakeem at gmail.com> said:
>> "Kurtis,
>> Which protocol will this chat/sms dialplan be compliant with ?.
>> Sip MESSAGE method , SIMPLE and XMPP are both widely adopted.
>> It would be great to add support for connectivity to external domains such
>> as
>> Gtalk, Yahoo Messenger etc.
>> Cheers,
>> AH"
>>
>> So far as my current architectural vision, any module implementing a
>> MESSAGE event should be able to trigger a dialplan action. My plan is
>> to start with Sofia, so it would support SIMPLE. Others should be
>> pretty easy as well.
>>
>> Lastly,
>> Anthony Minessale <anthony.minessale at gmail.com> said:
>> "This sounds redundant and intrusive.
>> We already have a concept of namespace within the chat username which
>> is used for routing.
>>
>> we prefix the namespace into the user id such as conf+1234 at domain.com
>> This allows conference to get messages from sip, xmpp or any other
>> protocol that supports chat.
>>
>> When a from user on chat contains +:
>>
>> each module just parses the inbound chat message, cuts off the proto
>> "everything up to the first +" in this case conf
>> and routes it to the module registered to that namespace.
>>
>> Right now conference will advertise it's presence and receive chat of
>> the word list and respond with the list despite the protocol used, eg
>> sip, jabber etc.
>>
>> Also chat messages are routed to individual calls when they are
>> associated with an active channel allowing you to use chat inside IVR.
>>
>> I recommend trying to understand all of this before proposing drastic
>> changes that will regress all of this functionality."
>>
>> I need a programmatic layer for routing messages. What's currently in
>> FS is (seemingly) unable to do this. I'm aware that the current system
>> routes messages to users correctly; I don't think it enables any sort
>> of interesting programmatic access. For instance, building a system
>> that receives a message and immediately starts a call is not something
>> I think can be currently done. The dialplan abstraction is
>> well-understood by FS users, and an extension I think could provide a
>> lot of value.
>>
>> I don't think these changes will regress any functionality either. I
>> proposed adding a configuration flag that switched between behaviors,
>> so that existing functionality won't be modified. Lastly, sending this
>> email was part of me trying to understand the current system. Thanks
>> for the direction.
>>
>> If I'm wrong, and there's a simple way to provide programmatic access,
>> I'd love to hear about it. That's a lot less work for me!
>>
>> On Sun, Jul 3, 2011 at 4:49 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
>> wrote:
>> > Hello Freeswitch-dev!
>> >
>> > I've recently begin modifying Sofia to support a chat "dialplan", very
>> > similar to the voice dialplan. Basically, a chat/sms will come in at a
>> > certain number/address and be routed via the dialplan. My current plan
>> > for doing this is as follows:
>> >
>> > 1) Add message events to sofia: I want sofia (and eventually all chat
>> > clients) to send message events through FS's event framework when an
>> > event arrives.
>> > 2) Remove message routing from Sofia: Sofia should not be routing
>> > messages itself. Instead, it should just generate the event saying a
>> > message has arrived and someone else should do the routing (namely my
>> > dialplan)
>> > 3) Implement a simple chat dialplan module that listens for message
>> > events and routes them according to an XML dialplan.
>> >
>> > I have some questions for the FS developer community.
>> >
>> > First, building my own module is conceptually simple. Modifying Sofia
>> > is not. What's the best way to go about making these changes? I could
>> > submit extension tickets (e.g., a configuration variable that adds
>> > messages and one that removes routing), or submit patches, or both. I
>> > think this could be broadly useful, and I want to try to contribute to
>> > the community.
>> >
>> > Secondly, ?I'm having a little difficulty finding the place where
>> > messages are routed in sofia. That's probably the place to insert both
>> > 1 & 2. Any pointers would be appreciated.
>> >
>> > Thanks!
>> >
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Wed Jul  6 09:32:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jul 2011 22:32:16 -0700
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
	<CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>
Message-ID: <CAKzWOxWFtGYsswjyHne29zghL_O+Acmr8+LGG4bCYJpwvOdaQg@mail.gmail.com>

On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>wrote:

> I don't think so. I set the ESL to log all events (/event all) from
> the command line and received no events when sofia received SIMPLE
> messages. Is that a bug? I had assumed it's intentional.
>
I'm 99% sure that those messages never make it up into FS from Sofia. I'd go
look into the code but Sofia is really scary and there's no amount of Scooby
Snacks you could give me to convince me to wander into that creepy old code.
:)
-MC
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From anthony.minessale at gmail.com  Wed Jul  6 09:49:37 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jul 2011 00:49:37 -0500
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CAKzWOxWFtGYsswjyHne29zghL_O+Acmr8+LGG4bCYJpwvOdaQg@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
	<CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>
	<CAKzWOxWFtGYsswjyHne29zghL_O+Acmr8+LGG4bCYJpwvOdaQg@mail.gmail.com>
Message-ID: <CAKbxfG_R2GYs1TfABVpTEFY_zBDD51xguEp5jkAqhkbPyt_h2g@mail.gmail.com>

Read my response and my mod conference example again.
You can register a chat callback from any module bound to a particular
namespace.......
On Jul 6, 2011 12:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu
>wrote:
>
>> I don't think so. I set the ESL to log all events (/event all) from
>> the command line and received no events when sofia received SIMPLE
>> messages. Is that a bug? I had assumed it's intentional.
>>
> I'm 99% sure that those messages never make it up into FS from Sofia. I'd
go
> look into the code but Sofia is really scary and there's no amount of
Scooby
> Snacks you could give me to convince me to wander into that creepy old
code.
> :)
> -MC
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From kheimerl at cs.berkeley.edu  Wed Jul  6 09:56:06 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Tue, 5 Jul 2011 22:56:06 -0700
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CAKbxfG_R2GYs1TfABVpTEFY_zBDD51xguEp5jkAqhkbPyt_h2g@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
	<CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>
	<CAKzWOxWFtGYsswjyHne29zghL_O+Acmr8+LGG4bCYJpwvOdaQg@mail.gmail.com>
	<CAKbxfG_R2GYs1TfABVpTEFY_zBDD51xguEp5jkAqhkbPyt_h2g@mail.gmail.com>
Message-ID: <CACyT-3mXz0qQ0KS_BKp06jNeVfJHXr2=T2-SwpzWE-MVj7=S=Q@mail.gmail.com>

I think I understand it. Basically, a client can prepend a +code that
causes the message to be routed to a specific module. I don't think
that fulfills my requirements. This is fine for basic routing, but
it's not a very good programmable environment for chat applications,
unless I'm missing something, which is always possible.

If I am, can you direct me to anything that explains it in a little
more depth than your email?

On Tue, Jul 5, 2011 at 10:49 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Read my response and my mod conference example again.
> You can register a chat callback from any module bound to a particular
> namespace.......
>
> On Jul 6, 2011 12:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu>wrote:
>>
>>> I don't think so. I set the ESL to log all events (/event all) from
>>> the command line and received no events when sofia received SIMPLE
>>> messages. Is that a bug? I had assumed it's intentional.
>>>
>> I'm 99% sure that those messages never make it up into FS from Sofia. I'd
>> go
>> look into the code but Sofia is really scary and there's no amount of
>> Scooby
>> Snacks you could give me to convince me to wander into that creepy old
>> code.
>> :)
>> -MC
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Wed Jul  6 10:25:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jul 2011 23:25:16 -0700
Subject: [Freeswitch-dev] Trouble with bind_digit_action and
	exec:execute_extension
Message-ID: <CAKzWOxV0UT=ReDs2Z=xuOsywjHQ3AR2o7ct2viMCWPyE8yY8Aw@mail.gmail.com>

Hello,

I know that some have recently reported having trouble with
bind_digit_action and exec:execute_extension. I have not been able to
reproduce any of the reported symptoms. I've produced a simple test dialplan
that you can drop into conf/dialplan/default/ and just reloadxml for a quick
test. I've attached it and also pasted it here for reference. If you've been
having BDA trouble please try this out and report back. Be sure to use
pastebin.freeswitch.org for reporting your logs.

-MC

Dialplan:
  <extension name="BDA Test">


    <condition field="destination_number" expression="^77(\d\d\d\d)$">


      <action application="answer"/>


      <action application="bind_digit_action"
data="test1,*1,exec:execute_extension,BDA_TARGET_1 XML default"/>

      <action application="bind_digit_action"
data="test1,*2,exec:execute_extension,BDA_TARGET_2 XML default"/>

      <action application="bind_digit_action"
data="test1,*3,exec:execute_extension,BDA_TARGET_3 XML default"/>

      <action application="bind_digit_action"
data="test1,*4,exec:execute_extension,BDA_TARGET_4 XML default"/>

      <action application="bind_digit_action"
data="test1,*5,exec:execute_extension,BDA_TARGET_5 XML default"/>

      <action application="bind_digit_action"
data="test1,*6,exec:execute_extension,BDA_TARGET_6 XML default"/>

      <action application="digit_action_set_realm" data="test1"/>


      <action application="playback" data="${hold_music}"/>


    </condition>


  </extension>





  <extension name="BDA Target">


    <condition field="destination_number" expression="^BDA_TARGET_(\d)$">


      <action application="say" data="en number pronounced $1"/>


    </condition>


  </extension>

Just dial 77xxxx and then press *1, *2, etc. You will hear the number
voiced, i.e. *1 will say "one", *2 will say "two" and so forth.
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From steveayre at gmail.com  Wed Jul  6 12:27:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 6 Jul 2011 09:27:23 +0100
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
Message-ID: <CAFiqYukcFuuwH-NMsou-q=cHwyTkwzyprGBoKbNtMeKB-z_woA@mail.gmail.com>

>
> I don't quite understand what you're saying here. Individual modules
> (e.g., Sofia) would register a different dialplan? Is that how it
> currently works? I thought, if they selected an XML dialplan in the
> conf, they all used the same one. I'm obviously confused, so please
> explain in more depth!
>

There's an interface so a module can register a dialplan handler.
mod_dialplan_xml provides the one most commonly used. Endpoint modules like
Sofia don't have a anything to do with the dialplan interface, that's the
job of another module. Other modules that implement dialplan handlers
include mod_dialplan_asterisk, mod_dialplan_directory, mod_yaml, mod_lua,
mod_enum, mod_osp, mod_lcr. Most people just use the XML one though. The
"XML" you often see in documentation for things like transfer actually is
you specifying you want to use the mod_dialplan_xml one. The other modules
use different dialplan names such as asterisk, directory, lua, lcr etc.

-Steve



On 6 July 2011 03:31, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:

> I forgot to disable digest mode, and so I have 5 emails to respond to
> in 2 received mails. Instead, I'm going to address the highlights
> here:
>
> Steven Ayre <steveayre at gmail.com> said:
> "Have you thought of making it a patch to add a new API interface so
> that modules can register chat dialplans, similar to how
> mod_dialplan_xml works?
>
> -Steve"
>
> I don't quite understand what you're saying here. Individual modules
> (e.g., Sofia) would register a different dialplan? Is that how it
> currently works? I thought, if they selected an XML dialplan in the
> conf, they all used the same one. I'm obviously confused, so please
> explain in more depth!
>
> "Abdul Hakeem" <alhakeem at gmail.com> said:
> "Kurtis,
> Which protocol will this chat/sms dialplan be compliant with ?.
> Sip MESSAGE method , SIMPLE and XMPP are both widely adopted.
> It would be great to add support for connectivity to external domains such
> as
> Gtalk, Yahoo Messenger etc.
> Cheers,
> AH"
>
> So far as my current architectural vision, any module implementing a
> MESSAGE event should be able to trigger a dialplan action. My plan is
> to start with Sofia, so it would support SIMPLE. Others should be
> pretty easy as well.
>
> Lastly,
> Anthony Minessale <anthony.minessale at gmail.com> said:
> "This sounds redundant and intrusive.
> We already have a concept of namespace within the chat username which
> is used for routing.
>
> we prefix the namespace into the user id such as conf+1234 at domain.com
> This allows conference to get messages from sip, xmpp or any other
> protocol that supports chat.
>
> When a from user on chat contains +:
>
> each module just parses the inbound chat message, cuts off the proto
> "everything up to the first +" in this case conf
> and routes it to the module registered to that namespace.
>
> Right now conference will advertise it's presence and receive chat of
> the word list and respond with the list despite the protocol used, eg
> sip, jabber etc.
>
> Also chat messages are routed to individual calls when they are
> associated with an active channel allowing you to use chat inside IVR.
>
> I recommend trying to understand all of this before proposing drastic
> changes that will regress all of this functionality."
>
> I need a programmatic layer for routing messages. What's currently in
> FS is (seemingly) unable to do this. I'm aware that the current system
> routes messages to users correctly; I don't think it enables any sort
> of interesting programmatic access. For instance, building a system
> that receives a message and immediately starts a call is not something
> I think can be currently done. The dialplan abstraction is
> well-understood by FS users, and an extension I think could provide a
> lot of value.
>
> I don't think these changes will regress any functionality either. I
> proposed adding a configuration flag that switched between behaviors,
> so that existing functionality won't be modified. Lastly, sending this
> email was part of me trying to understand the current system. Thanks
> for the direction.
>
> If I'm wrong, and there's a simple way to provide programmatic access,
> I'd love to hear about it. That's a lot less work for me!
>
> On Sun, Jul 3, 2011 at 4:49 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
> wrote:
> > Hello Freeswitch-dev!
> >
> > I've recently begin modifying Sofia to support a chat "dialplan", very
> > similar to the voice dialplan. Basically, a chat/sms will come in at a
> > certain number/address and be routed via the dialplan. My current plan
> > for doing this is as follows:
> >
> > 1) Add message events to sofia: I want sofia (and eventually all chat
> > clients) to send message events through FS's event framework when an
> > event arrives.
> > 2) Remove message routing from Sofia: Sofia should not be routing
> > messages itself. Instead, it should just generate the event saying a
> > message has arrived and someone else should do the routing (namely my
> > dialplan)
> > 3) Implement a simple chat dialplan module that listens for message
> > events and routes them according to an XML dialplan.
> >
> > I have some questions for the FS developer community.
> >
> > First, building my own module is conceptually simple. Modifying Sofia
> > is not. What's the best way to go about making these changes? I could
> > submit extension tickets (e.g., a configuration variable that adds
> > messages and one that removes routing), or submit patches, or both. I
> > think this could be broadly useful, and I want to try to contribute to
> > the community.
> >
> > Secondly,  I'm having a little difficulty finding the place where
> > messages are routed in sofia. That's probably the place to insert both
> > 1 & 2. Any pointers would be appreciated.
> >
> > Thanks!
> >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Wed Jul  6 12:27:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 6 Jul 2011 09:27:47 +0100
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CAFiqYukcFuuwH-NMsou-q=cHwyTkwzyprGBoKbNtMeKB-z_woA@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CAFiqYukcFuuwH-NMsou-q=cHwyTkwzyprGBoKbNtMeKB-z_woA@mail.gmail.com>
Message-ID: <CAFiqYumyubgGqC6nRtep+F41Jgdgx3a=_2OGj23xJEN94+WJpw@mail.gmail.com>

(grep for SWITCH_ADD_DIALPLAN if you're interested)



On 6 July 2011 09:27, Steven Ayre <steveayre at gmail.com> wrote:

> I don't quite understand what you're saying here. Individual modules
>> (e.g., Sofia) would register a different dialplan? Is that how it
>> currently works? I thought, if they selected an XML dialplan in the
>> conf, they all used the same one. I'm obviously confused, so please
>> explain in more depth!
>>
>
> There's an interface so a module can register a dialplan handler.
> mod_dialplan_xml provides the one most commonly used. Endpoint modules like
> Sofia don't have a anything to do with the dialplan interface, that's the
> job of another module. Other modules that implement dialplan handlers
> include mod_dialplan_asterisk, mod_dialplan_directory, mod_yaml, mod_lua,
> mod_enum, mod_osp, mod_lcr. Most people just use the XML one though. The
> "XML" you often see in documentation for things like transfer actually is
> you specifying you want to use the mod_dialplan_xml one. The other modules
> use different dialplan names such as asterisk, directory, lua, lcr etc.
>
> -Steve
>
>
>
>
> On 6 July 2011 03:31, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>
>> I forgot to disable digest mode, and so I have 5 emails to respond to
>> in 2 received mails. Instead, I'm going to address the highlights
>> here:
>>
>> Steven Ayre <steveayre at gmail.com> said:
>> "Have you thought of making it a patch to add a new API interface so
>> that modules can register chat dialplans, similar to how
>> mod_dialplan_xml works?
>>
>> -Steve"
>>
>> I don't quite understand what you're saying here. Individual modules
>> (e.g., Sofia) would register a different dialplan? Is that how it
>> currently works? I thought, if they selected an XML dialplan in the
>> conf, they all used the same one. I'm obviously confused, so please
>> explain in more depth!
>>
>> "Abdul Hakeem" <alhakeem at gmail.com> said:
>> "Kurtis,
>> Which protocol will this chat/sms dialplan be compliant with ?.
>> Sip MESSAGE method , SIMPLE and XMPP are both widely adopted.
>> It would be great to add support for connectivity to external domains such
>> as
>> Gtalk, Yahoo Messenger etc.
>> Cheers,
>> AH"
>>
>> So far as my current architectural vision, any module implementing a
>> MESSAGE event should be able to trigger a dialplan action. My plan is
>> to start with Sofia, so it would support SIMPLE. Others should be
>> pretty easy as well.
>>
>> Lastly,
>> Anthony Minessale <anthony.minessale at gmail.com> said:
>> "This sounds redundant and intrusive.
>> We already have a concept of namespace within the chat username which
>> is used for routing.
>>
>> we prefix the namespace into the user id such as conf+1234 at domain.com
>> This allows conference to get messages from sip, xmpp or any other
>> protocol that supports chat.
>>
>> When a from user on chat contains +:
>>
>> each module just parses the inbound chat message, cuts off the proto
>> "everything up to the first +" in this case conf
>> and routes it to the module registered to that namespace.
>>
>> Right now conference will advertise it's presence and receive chat of
>> the word list and respond with the list despite the protocol used, eg
>> sip, jabber etc.
>>
>> Also chat messages are routed to individual calls when they are
>> associated with an active channel allowing you to use chat inside IVR.
>>
>> I recommend trying to understand all of this before proposing drastic
>> changes that will regress all of this functionality."
>>
>> I need a programmatic layer for routing messages. What's currently in
>> FS is (seemingly) unable to do this. I'm aware that the current system
>> routes messages to users correctly; I don't think it enables any sort
>> of interesting programmatic access. For instance, building a system
>> that receives a message and immediately starts a call is not something
>> I think can be currently done. The dialplan abstraction is
>> well-understood by FS users, and an extension I think could provide a
>> lot of value.
>>
>> I don't think these changes will regress any functionality either. I
>> proposed adding a configuration flag that switched between behaviors,
>> so that existing functionality won't be modified. Lastly, sending this
>> email was part of me trying to understand the current system. Thanks
>> for the direction.
>>
>> If I'm wrong, and there's a simple way to provide programmatic access,
>> I'd love to hear about it. That's a lot less work for me!
>>
>> On Sun, Jul 3, 2011 at 4:49 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
>> wrote:
>> > Hello Freeswitch-dev!
>> >
>> > I've recently begin modifying Sofia to support a chat "dialplan", very
>> > similar to the voice dialplan. Basically, a chat/sms will come in at a
>> > certain number/address and be routed via the dialplan. My current plan
>> > for doing this is as follows:
>> >
>> > 1) Add message events to sofia: I want sofia (and eventually all chat
>> > clients) to send message events through FS's event framework when an
>> > event arrives.
>> > 2) Remove message routing from Sofia: Sofia should not be routing
>> > messages itself. Instead, it should just generate the event saying a
>> > message has arrived and someone else should do the routing (namely my
>> > dialplan)
>> > 3) Implement a simple chat dialplan module that listens for message
>> > events and routes them according to an XML dialplan.
>> >
>> > I have some questions for the FS developer community.
>> >
>> > First, building my own module is conceptually simple. Modifying Sofia
>> > is not. What's the best way to go about making these changes? I could
>> > submit extension tickets (e.g., a configuration variable that adds
>> > messages and one that removes routing), or submit patches, or both. I
>> > think this could be broadly useful, and I want to try to contribute to
>> > the community.
>> >
>> > Secondly,  I'm having a little difficulty finding the place where
>> > messages are routed in sofia. That's probably the place to insert both
>> > 1 & 2. Any pointers would be appreciated.
>> >
>> > Thanks!
>> >
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
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From steveayre at gmail.com  Wed Jul  6 12:29:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 6 Jul 2011 09:29:46 +0100
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
Message-ID: <CAFiqYunqrW8Yh14o6ozkTvmqaP-00DZcbjEWoepeBOW+6CNR3g@mail.gmail.com>

You can also write a module to load in FS that can listen to the same
events, so programmatically may be able to handle them within FS there.

-Steve


2011/7/6 Jo?o Mesquita <jmesquita at freeswitch.org>

> I believe that these messages show up as events on ESL, don't they?
> Therefore you can do whatever from there...
>
> Regards,
> Jo?o Mesquita
>
>
>
>
> On Tue, Jul 5, 2011 at 11:31 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>wrote:
>
>> I forgot to disable digest mode, and so I have 5 emails to respond to
>> in 2 received mails. Instead, I'm going to address the highlights
>> here:
>>
>> Steven Ayre <steveayre at gmail.com> said:
>> "Have you thought of making it a patch to add a new API interface so
>> that modules can register chat dialplans, similar to how
>> mod_dialplan_xml works?
>>
>> -Steve"
>>
>> I don't quite understand what you're saying here. Individual modules
>> (e.g., Sofia) would register a different dialplan? Is that how it
>> currently works? I thought, if they selected an XML dialplan in the
>> conf, they all used the same one. I'm obviously confused, so please
>> explain in more depth!
>>
>> "Abdul Hakeem" <alhakeem at gmail.com> said:
>> "Kurtis,
>> Which protocol will this chat/sms dialplan be compliant with ?.
>> Sip MESSAGE method , SIMPLE and XMPP are both widely adopted.
>> It would be great to add support for connectivity to external domains such
>> as
>> Gtalk, Yahoo Messenger etc.
>> Cheers,
>> AH"
>>
>> So far as my current architectural vision, any module implementing a
>> MESSAGE event should be able to trigger a dialplan action. My plan is
>> to start with Sofia, so it would support SIMPLE. Others should be
>> pretty easy as well.
>>
>> Lastly,
>> Anthony Minessale <anthony.minessale at gmail.com> said:
>> "This sounds redundant and intrusive.
>> We already have a concept of namespace within the chat username which
>> is used for routing.
>>
>> we prefix the namespace into the user id such as conf+1234 at domain.com
>> This allows conference to get messages from sip, xmpp or any other
>> protocol that supports chat.
>>
>> When a from user on chat contains +:
>>
>> each module just parses the inbound chat message, cuts off the proto
>> "everything up to the first +" in this case conf
>> and routes it to the module registered to that namespace.
>>
>> Right now conference will advertise it's presence and receive chat of
>> the word list and respond with the list despite the protocol used, eg
>> sip, jabber etc.
>>
>> Also chat messages are routed to individual calls when they are
>> associated with an active channel allowing you to use chat inside IVR.
>>
>> I recommend trying to understand all of this before proposing drastic
>> changes that will regress all of this functionality."
>>
>> I need a programmatic layer for routing messages. What's currently in
>> FS is (seemingly) unable to do this. I'm aware that the current system
>> routes messages to users correctly; I don't think it enables any sort
>> of interesting programmatic access. For instance, building a system
>> that receives a message and immediately starts a call is not something
>> I think can be currently done. The dialplan abstraction is
>> well-understood by FS users, and an extension I think could provide a
>> lot of value.
>>
>> I don't think these changes will regress any functionality either. I
>> proposed adding a configuration flag that switched between behaviors,
>> so that existing functionality won't be modified. Lastly, sending this
>> email was part of me trying to understand the current system. Thanks
>> for the direction.
>>
>> If I'm wrong, and there's a simple way to provide programmatic access,
>> I'd love to hear about it. That's a lot less work for me!
>>
>> On Sun, Jul 3, 2011 at 4:49 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
>> wrote:
>> > Hello Freeswitch-dev!
>> >
>> > I've recently begin modifying Sofia to support a chat "dialplan", very
>> > similar to the voice dialplan. Basically, a chat/sms will come in at a
>> > certain number/address and be routed via the dialplan. My current plan
>> > for doing this is as follows:
>> >
>> > 1) Add message events to sofia: I want sofia (and eventually all chat
>> > clients) to send message events through FS's event framework when an
>> > event arrives.
>> > 2) Remove message routing from Sofia: Sofia should not be routing
>> > messages itself. Instead, it should just generate the event saying a
>> > message has arrived and someone else should do the routing (namely my
>> > dialplan)
>> > 3) Implement a simple chat dialplan module that listens for message
>> > events and routes them according to an XML dialplan.
>> >
>> > I have some questions for the FS developer community.
>> >
>> > First, building my own module is conceptually simple. Modifying Sofia
>> > is not. What's the best way to go about making these changes? I could
>> > submit extension tickets (e.g., a configuration variable that adds
>> > messages and one that removes routing), or submit patches, or both. I
>> > think this could be broadly useful, and I want to try to contribute to
>> > the community.
>> >
>> > Secondly,  I'm having a little difficulty finding the place where
>> > messages are routed in sofia. That's probably the place to insert both
>> > 1 & 2. Any pointers would be appreciated.
>> >
>> > Thanks!
>> >
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From kheimerl at cs.berkeley.edu  Wed Jul  6 13:52:18 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Wed, 6 Jul 2011 02:52:18 -0700
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CAFiqYunqrW8Yh14o6ozkTvmqaP-00DZcbjEWoepeBOW+6CNR3g@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
	<CAFiqYunqrW8Yh14o6ozkTvmqaP-00DZcbjEWoepeBOW+6CNR3g@mail.gmail.com>
Message-ID: <CACyT-3=0RDrTfmEp3YP5zZr-Vm89gPPTF388n53QiVKrubD5rQ@mail.gmail.com>

Basically, I expect my module to listen for message events and forward
them to a dialplan (of any kind, likely using this API).

Thanks for the direction!

On Wed, Jul 6, 2011 at 1:29 AM, Steven Ayre <steveayre at gmail.com> wrote:
> You can also write a module to load in FS that can listen to the same
> events, so programmatically may be able to handle them within FS there.
>
> -Steve
>
>
> 2011/7/6 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> I believe that these messages show up as events on ESL, don't they?
>> Therefore you can do whatever from there...
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>> On Tue, Jul 5, 2011 at 11:31 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
>> wrote:
>>>
>>> I forgot to disable digest mode, and so I have 5 emails to respond to
>>> in 2 received mails. Instead, I'm going to address the highlights
>>> here:
>>>
>>> Steven Ayre <steveayre at gmail.com> said:
>>> "Have you thought of making it a patch to add a new API interface so
>>> that modules can register chat dialplans, similar to how
>>> mod_dialplan_xml works?
>>>
>>> -Steve"
>>>
>>> I don't quite understand what you're saying here. Individual modules
>>> (e.g., Sofia) would register a different dialplan? Is that how it
>>> currently works? I thought, if they selected an XML dialplan in the
>>> conf, they all used the same one. I'm obviously confused, so please
>>> explain in more depth!
>>>
>>> "Abdul Hakeem" <alhakeem at gmail.com> said:
>>> "Kurtis,
>>> Which protocol will this chat/sms dialplan be compliant with ?.
>>> Sip MESSAGE method , SIMPLE and XMPP are both widely adopted.
>>> It would be great to add support for connectivity to external domains
>>> such as
>>> Gtalk, Yahoo Messenger etc.
>>> Cheers,
>>> AH"
>>>
>>> So far as my current architectural vision, any module implementing a
>>> MESSAGE event should be able to trigger a dialplan action. My plan is
>>> to start with Sofia, so it would support SIMPLE. Others should be
>>> pretty easy as well.
>>>
>>> Lastly,
>>> Anthony Minessale <anthony.minessale at gmail.com> said:
>>> "This sounds redundant and intrusive.
>>> We already have a concept of namespace within the chat username which
>>> is used for routing.
>>>
>>> we prefix the namespace into the user id such as conf+1234 at domain.com
>>> This allows conference to get messages from sip, xmpp or any other
>>> protocol that supports chat.
>>>
>>> When a from user on chat contains +:
>>>
>>> each module just parses the inbound chat message, cuts off the proto
>>> "everything up to the first +" in this case conf
>>> and routes it to the module registered to that namespace.
>>>
>>> Right now conference will advertise it's presence and receive chat of
>>> the word list and respond with the list despite the protocol used, eg
>>> sip, jabber etc.
>>>
>>> Also chat messages are routed to individual calls when they are
>>> associated with an active channel allowing you to use chat inside IVR.
>>>
>>> I recommend trying to understand all of this before proposing drastic
>>> changes that will regress all of this functionality."
>>>
>>> I need a programmatic layer for routing messages. What's currently in
>>> FS is (seemingly) unable to do this. I'm aware that the current system
>>> routes messages to users correctly; I don't think it enables any sort
>>> of interesting programmatic access. For instance, building a system
>>> that receives a message and immediately starts a call is not something
>>> I think can be currently done. The dialplan abstraction is
>>> well-understood by FS users, and an extension I think could provide a
>>> lot of value.
>>>
>>> I don't think these changes will regress any functionality either. I
>>> proposed adding a configuration flag that switched between behaviors,
>>> so that existing functionality won't be modified. Lastly, sending this
>>> email was part of me trying to understand the current system. Thanks
>>> for the direction.
>>>
>>> If I'm wrong, and there's a simple way to provide programmatic access,
>>> I'd love to hear about it. That's a lot less work for me!
>>>
>>> On Sun, Jul 3, 2011 at 4:49 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
>>> wrote:
>>> > Hello Freeswitch-dev!
>>> >
>>> > I've recently begin modifying Sofia to support a chat "dialplan", very
>>> > similar to the voice dialplan. Basically, a chat/sms will come in at a
>>> > certain number/address and be routed via the dialplan. My current plan
>>> > for doing this is as follows:
>>> >
>>> > 1) Add message events to sofia: I want sofia (and eventually all chat
>>> > clients) to send message events through FS's event framework when an
>>> > event arrives.
>>> > 2) Remove message routing from Sofia: Sofia should not be routing
>>> > messages itself. Instead, it should just generate the event saying a
>>> > message has arrived and someone else should do the routing (namely my
>>> > dialplan)
>>> > 3) Implement a simple chat dialplan module that listens for message
>>> > events and routes them according to an XML dialplan.
>>> >
>>> > I have some questions for the FS developer community.
>>> >
>>> > First, building my own module is conceptually simple. Modifying Sofia
>>> > is not. What's the best way to go about making these changes? I could
>>> > submit extension tickets (e.g., a configuration variable that adds
>>> > messages and one that removes routing), or submit patches, or both. I
>>> > think this could be broadly useful, and I want to try to contribute to
>>> > the community.
>>> >
>>> > Secondly, ?I'm having a little difficulty finding the place where
>>> > messages are routed in sofia. That's probably the place to insert both
>>> > 1 & 2. Any pointers would be appreciated.
>>> >
>>> > Thanks!
>>> >
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From dome at tel.co.th  Wed Jul  6 13:58:41 2011
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 6 Jul 2011 16:58:41 +0700
Subject: [Freeswitch-dev] api_hangup_hook and channel variable
In-Reply-To: <CAKzWOxWaMG7sR4oMzehLBPEd6Lo38UtB8r9iu2DyLyRHajbc7w@mail.gmail.com>
References: <CAMFECSAxxu51JE2nNYPTThgn=L-AYHTqEqXC6zWSAHfjNbjSRQ@mail.gmail.com>
	<CAKzWOxWaMG7sR4oMzehLBPEd6Lo38UtB8r9iu2DyLyRHajbc7w@mail.gmail.com>
Message-ID: <CAMFECSAOeFeZFL6ia9Bcyzhoz6U+Nw7p94pWnVKSpgdmDpKTHw@mail.gmail.com>

2011/7/5 Michael Collins <msc at freeswitch.org>:
> Once you are in report.lua the channel is already hung up. To the best of my
> knowledge there is no way to add/manipulate channel variables in a hangup
> hook since you really just have a static copy of the channel variables.
> What is it that you are trying to do?
my system do realtime billing like a mod_nibblebill
but my module deduct balance by memcache protocol (balance store in
membase server)
so sometime i want to give them bonus (increase balance ) for some
call (like a over 10 min)
after hangup. i need to create new channel variable and  munipulate
some variable too.

Dome C.





> -MC
>
> On Tue, Jul 5, 2011 at 3:39 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> This is my lua script
>>
>> ? ? ? ? ? ?session:setVariable("session_in_hangup_hook", "true");
>> ? ? ? ? ? ?session:setVariable("api_hangup_hook", "lua
>> hangup/report.lua "..macct);
>> ? ? ? ? ? ?session:transfer("dialnow_wh", "XML", "wh");
>>
>> It's work fine before hangup i cal see session vation in report.lua
>> but my quession is how to manipulate or create new variable in
>> report.lua ??
>> i want to see variable in cdr
>>
>> BG
>>
>> Dome C.
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Wed Jul  6 19:46:43 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jul 2011 10:46:43 -0500
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3mXz0qQ0KS_BKp06jNeVfJHXr2=T2-SwpzWE-MVj7=S=Q@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
	<CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>
	<CAKzWOxWFtGYsswjyHne29zghL_O+Acmr8+LGG4bCYJpwvOdaQg@mail.gmail.com>
	<CAKbxfG_R2GYs1TfABVpTEFY_zBDD51xguEp5jkAqhkbPyt_h2g@mail.gmail.com>
	<CACyT-3mXz0qQ0KS_BKp06jNeVfJHXr2=T2-SwpzWE-MVj7=S=Q@mail.gmail.com>
Message-ID: <CAKbxfG_hmhmZzR7sCs1PW-2wF72auiJESD4pPkNNOjt7jenkRA@mail.gmail.com>

Are we talking about that ridiculous new chat concept in SIP where it
uses INVITES and dialogs?


On Wed, Jul 6, 2011 at 12:56 AM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> I think I understand it. Basically, a client can prepend a +code that
> causes the message to be routed to a specific module. I don't think
> that fulfills my requirements. This is fine for basic routing, but
> it's not a very good programmable environment for chat applications,
> unless I'm missing something, which is always possible.
>
> If I am, can you direct me to anything that explains it in a little
> more depth than your email?
>
> On Tue, Jul 5, 2011 at 10:49 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Read my response and my mod conference example again.
>> You can register a chat callback from any module bound to a particular
>> namespace.......
>>
>> On Jul 6, 2011 12:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>>> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl
>>> <kheimerl at cs.berkeley.edu>wrote:
>>>
>>>> I don't think so. I set the ESL to log all events (/event all) from
>>>> the command line and received no events when sofia received SIMPLE
>>>> messages. Is that a bug? I had assumed it's intentional.
>>>>
>>> I'm 99% sure that those messages never make it up into FS from Sofia. I'd
>>> go
>>> look into the code but Sofia is really scary and there's no amount of
>>> Scooby
>>> Snacks you could give me to convince me to wander into that creepy old
>>> code.
>>> :)
>>> -MC
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Wed Jul  6 21:13:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jul 2011 10:13:29 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxUrZtrXvJBWPWHX6vv-PPBQ=SCgS2CSwF60GU4_4TEU8Q@mail.gmail.com>

C'mon down!
http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_06

Lots of new stuff to talk about!
-MC
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From marketing at cluecon.com  Wed Jul  6 21:14:36 2011
From: marketing at cluecon.com (marketing at cluecon.com)
Date: Wed, 6 Jul 2011 17:14:36 +0000
Subject: [Freeswitch-dev] ClueCon News: Free OpenSIPS Training,
	Sponsor Logo - Last Chance, Party RSVP
Message-ID: <000001310071b10c-8645c79b-935a-49f0-8331-94a3f7a972d6-000000@email.amazonses.com>

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URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110706/a98bd9ed/attachment.html 

From msc at freeswitch.org  Wed Jul  6 21:27:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jul 2011 10:27:05 -0700
Subject: [Freeswitch-dev] Flex RTMP Client Added To Git Tree!
Message-ID: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>

I thought many of you would be interested in this story about the RTMP Flex
client:

http://www.freeswitch.org/node/332

The sample Flex client has been added to tree, so feel free to start
experimenting!

-Michael
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From kheimerl at cs.berkeley.edu  Thu Jul  7 03:14:34 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Wed, 6 Jul 2011 16:14:34 -0700
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CAKbxfG_hmhmZzR7sCs1PW-2wF72auiJESD4pPkNNOjt7jenkRA@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
	<CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>
	<CAKzWOxWFtGYsswjyHne29zghL_O+Acmr8+LGG4bCYJpwvOdaQg@mail.gmail.com>
	<CAKbxfG_R2GYs1TfABVpTEFY_zBDD51xguEp5jkAqhkbPyt_h2g@mail.gmail.com>
	<CACyT-3mXz0qQ0KS_BKp06jNeVfJHXr2=T2-SwpzWE-MVj7=S=Q@mail.gmail.com>
	<CAKbxfG_hmhmZzR7sCs1PW-2wF72auiJESD4pPkNNOjt7jenkRA@mail.gmail.com>
Message-ID: <CACyT-3=iy4G4ZxfG_N3iTgQZZSRLGi2D1RvX7U0SBXpgrAWVyA@mail.gmail.com>

Nope, normal SIP MESSAGE events, no invites.

I just need an easy place where an app can look up some variables in
the directory, check presence information, and start a call based on
an incoming chat event. I also need a system that interops with
existing SIP chat services. I don't think the +routing does these for
me.

On Wed, Jul 6, 2011 at 8:46 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Are we talking about that ridiculous new chat concept in SIP where it
> uses INVITES and dialogs?
>
>
> On Wed, Jul 6, 2011 at 12:56 AM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> I think I understand it. Basically, a client can prepend a +code that
>> causes the message to be routed to a specific module. I don't think
>> that fulfills my requirements. This is fine for basic routing, but
>> it's not a very good programmable environment for chat applications,
>> unless I'm missing something, which is always possible.
>>
>> If I am, can you direct me to anything that explains it in a little
>> more depth than your email?
>>
>> On Tue, Jul 5, 2011 at 10:49 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> Read my response and my mod conference example again.
>>> You can register a chat callback from any module bound to a particular
>>> namespace.......
>>>
>>> On Jul 6, 2011 12:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>>>> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl
>>>> <kheimerl at cs.berkeley.edu>wrote:
>>>>
>>>>> I don't think so. I set the ESL to log all events (/event all) from
>>>>> the command line and received no events when sofia received SIMPLE
>>>>> messages. Is that a bug? I had assumed it's intentional.
>>>>>
>>>> I'm 99% sure that those messages never make it up into FS from Sofia. I'd
>>>> go
>>>> look into the code but Sofia is really scary and there's no amount of
>>>> Scooby
>>>> Snacks you could give me to convince me to wander into that creepy old
>>>> code.
>>>> :)
>>>> -MC
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Thu Jul  7 04:23:16 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jul 2011 19:23:16 -0500
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3=iy4G4ZxfG_N3iTgQZZSRLGi2D1RvX7U0SBXpgrAWVyA@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
	<CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>
	<CAKzWOxWFtGYsswjyHne29zghL_O+Acmr8+LGG4bCYJpwvOdaQg@mail.gmail.com>
	<CAKbxfG_R2GYs1TfABVpTEFY_zBDD51xguEp5jkAqhkbPyt_h2g@mail.gmail.com>
	<CACyT-3mXz0qQ0KS_BKp06jNeVfJHXr2=T2-SwpzWE-MVj7=S=Q@mail.gmail.com>
	<CAKbxfG_hmhmZzR7sCs1PW-2wF72auiJESD4pPkNNOjt7jenkRA@mail.gmail.com>
	<CACyT-3=iy4G4ZxfG_N3iTgQZZSRLGi2D1RvX7U0SBXpgrAWVyA@mail.gmail.com>
Message-ID: <CAKbxfG_Sf9zy25CsKFbr5wJt3c_2kNGZs7benpzuqsfcWTqCBg@mail.gmail.com>

you control the user names on your system the prefix+ string is just
part of the username people would talk to.

for instance ext+ namespace is has auto presence built in so if you
add ext+1000 to your list it will auto advertise that it's online.

conf+ goes to the conference.

conf+888 at server.com will go to FS on server.com and route the chat
messages to the mod_conference.

so basically you are proposing to try to modify sofia and break the
other stuff  because you do not like the prefix string on the user
names but its designed to allow you to cross connect protocols, eg
mod_sofia has sip+ registered and mod_dingaling has jingle+ so you can
use jabber to pass sip+user at domain.com to your buddy list and have it
report accurate presence.

Basically you make a new module called mod_foo and register the foo namespace.
Then on your clients you subscribe to foo+1000 at server.com for instance.
Chat messages to that foo namespace will arrive in your code and you
have the option to reply to them completely agnostic of the protocol
sip, jabber etc.

You seem to be asking for advice but then ignoring it coming from the
author of all of the above who spent many man hours solving this
problem so it would be abstract.  so I'm lost for what else to tell
you.

Maybe we can confirm that you are simply taken aback by the idea of
starting all your usernames with foo+ and simply propose to unravel
everything in pursuit of removing it?





On Wed, Jul 6, 2011 at 6:14 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
> Nope, normal SIP MESSAGE events, no invites.
>
> I just need an easy place where an app can look up some variables in
> the directory, check presence information, and start a call based on
> an incoming chat event. I also need a system that interops with
> existing SIP chat services. I don't think the +routing does these for
> me.
>
> On Wed, Jul 6, 2011 at 8:46 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> Are we talking about that ridiculous new chat concept in SIP where it
>> uses INVITES and dialogs?
>>
>>
>> On Wed, Jul 6, 2011 at 12:56 AM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu> wrote:
>>> I think I understand it. Basically, a client can prepend a +code that
>>> causes the message to be routed to a specific module. I don't think
>>> that fulfills my requirements. This is fine for basic routing, but
>>> it's not a very good programmable environment for chat applications,
>>> unless I'm missing something, which is always possible.
>>>
>>> If I am, can you direct me to anything that explains it in a little
>>> more depth than your email?
>>>
>>> On Tue, Jul 5, 2011 at 10:49 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> Read my response and my mod conference example again.
>>>> You can register a chat callback from any module bound to a particular
>>>> namespace.......
>>>>
>>>> On Jul 6, 2011 12:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>>>>> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl
>>>>> <kheimerl at cs.berkeley.edu>wrote:
>>>>>
>>>>>> I don't think so. I set the ESL to log all events (/event all) from
>>>>>> the command line and received no events when sofia received SIMPLE
>>>>>> messages. Is that a bug? I had assumed it's intentional.
>>>>>>
>>>>> I'm 99% sure that those messages never make it up into FS from Sofia. I'd
>>>>> go
>>>>> look into the code but Sofia is really scary and there's no amount of
>>>>> Scooby
>>>>> Snacks you could give me to convince me to wander into that creepy old
>>>>> code.
>>>>> :)
>>>>> -MC
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From kheimerl at cs.berkeley.edu  Thu Jul  7 04:50:07 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Wed, 6 Jul 2011 17:50:07 -0700
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CAKbxfG_Sf9zy25CsKFbr5wJt3c_2kNGZs7benpzuqsfcWTqCBg@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
	<CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>
	<CAKzWOxWFtGYsswjyHne29zghL_O+Acmr8+LGG4bCYJpwvOdaQg@mail.gmail.com>
	<CAKbxfG_R2GYs1TfABVpTEFY_zBDD51xguEp5jkAqhkbPyt_h2g@mail.gmail.com>
	<CACyT-3mXz0qQ0KS_BKp06jNeVfJHXr2=T2-SwpzWE-MVj7=S=Q@mail.gmail.com>
	<CAKbxfG_hmhmZzR7sCs1PW-2wF72auiJESD4pPkNNOjt7jenkRA@mail.gmail.com>
	<CACyT-3=iy4G4ZxfG_N3iTgQZZSRLGi2D1RvX7U0SBXpgrAWVyA@mail.gmail.com>
	<CAKbxfG_Sf9zy25CsKFbr5wJt3c_2kNGZs7benpzuqsfcWTqCBg@mail.gmail.com>
Message-ID: <CACyT-3mwg8GMuDgkGQ+593vsqNne4aVX3BXKjMzBgrBnEQwDyw@mail.gmail.com>

Let me repeat myself: the changes I am proposing will be backwards
compatible. Nothing will be broken.

The dialplan abstraction is tangential to this whole discussion. I can
build a mod foo which runs the dialplan, and have all messages routed
to it via the + mechanism. This will probably be my initial version of
the system. I understand the work you put into this, and that your
decisions were well-reasoned for the use cases you envisioned (e.g.,
conference calls), but the abstraction isn't working for me. This
system is just a piece, interoperating with a variety of other
systems. Attaching a specific module tag, used by just one piece of
one program, to a username in use throughout the entire system is not
an elegant solution. This means I have to put a proxy in front of FS,
adding this + tag to every message coming in.

I'm here and open to any solutions you might have to make the existing
solution work for me. Fundamentally, I can't just shove a +tag onto
every incoming message in any elegant way, so far as I know. I think
that means this entire line of reasoning is just broken for me. Don't
get me wrong here, I would be really really happy if I didn't have to
touch sofia to solve this, it's really hard to grok. It's also a lot
less work for me.

So far, I've proposed one alternate method for doing this. I want
sofia to generate events and not route messages itself. Then I can
listen to these perform whatever actions I need to. This may not work,
and I'm open to implementing any other architectural changes you think
would enable my use case. You're clearly the guy who knows what to do,
and your advice is invaluable.

On Wed, Jul 6, 2011 at 5:23 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you control the user names on your system the prefix+ string is just
> part of the username people would talk to.
>
> for instance ext+ namespace is has auto presence built in so if you
> add ext+1000 to your list it will auto advertise that it's online.
>
> conf+ goes to the conference.
>
> conf+888 at server.com will go to FS on server.com and route the chat
> messages to the mod_conference.
>
> so basically you are proposing to try to modify sofia and break the
> other stuff ?because you do not like the prefix string on the user
> names but its designed to allow you to cross connect protocols, eg
> mod_sofia has sip+ registered and mod_dingaling has jingle+ so you can
> use jabber to pass sip+user at domain.com to your buddy list and have it
> report accurate presence.
>
> Basically you make a new module called mod_foo and register the foo namespace.
> Then on your clients you subscribe to foo+1000 at server.com for instance.
> Chat messages to that foo namespace will arrive in your code and you
> have the option to reply to them completely agnostic of the protocol
> sip, jabber etc.
>
> You seem to be asking for advice but then ignoring it coming from the
> author of all of the above who spent many man hours solving this
> problem so it would be abstract. ?so I'm lost for what else to tell
> you.
>
> Maybe we can confirm that you are simply taken aback by the idea of
> starting all your usernames with foo+ and simply propose to unravel
> everything in pursuit of removing it?
>
>
>
>
>
> On Wed, Jul 6, 2011 at 6:14 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>> Nope, normal SIP MESSAGE events, no invites.
>>
>> I just need an easy place where an app can look up some variables in
>> the directory, check presence information, and start a call based on
>> an incoming chat event. I also need a system that interops with
>> existing SIP chat services. I don't think the +routing does these for
>> me.
>>
>> On Wed, Jul 6, 2011 at 8:46 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> Are we talking about that ridiculous new chat concept in SIP where it
>>> uses INVITES and dialogs?
>>>
>>>
>>> On Wed, Jul 6, 2011 at 12:56 AM, Kurtis Heimerl
>>> <kheimerl at cs.berkeley.edu> wrote:
>>>> I think I understand it. Basically, a client can prepend a +code that
>>>> causes the message to be routed to a specific module. I don't think
>>>> that fulfills my requirements. This is fine for basic routing, but
>>>> it's not a very good programmable environment for chat applications,
>>>> unless I'm missing something, which is always possible.
>>>>
>>>> If I am, can you direct me to anything that explains it in a little
>>>> more depth than your email?
>>>>
>>>> On Tue, Jul 5, 2011 at 10:49 PM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>>> Read my response and my mod conference example again.
>>>>> You can register a chat callback from any module bound to a particular
>>>>> namespace.......
>>>>>
>>>>> On Jul 6, 2011 12:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>>>>>> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl
>>>>>> <kheimerl at cs.berkeley.edu>wrote:
>>>>>>
>>>>>>> I don't think so. I set the ESL to log all events (/event all) from
>>>>>>> the command line and received no events when sofia received SIMPLE
>>>>>>> messages. Is that a bug? I had assumed it's intentional.
>>>>>>>
>>>>>> I'm 99% sure that those messages never make it up into FS from Sofia. I'd
>>>>>> go
>>>>>> look into the code but Sofia is really scary and there's no amount of
>>>>>> Scooby
>>>>>> Snacks you could give me to convince me to wander into that creepy old
>>>>>> code.
>>>>>> :)
>>>>>> -MC
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-dev mailing list
>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Thu Jul  7 05:21:43 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jul 2011 20:21:43 -0500
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CACyT-3mwg8GMuDgkGQ+593vsqNne4aVX3BXKjMzBgrBnEQwDyw@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
	<CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>
	<CAKzWOxWFtGYsswjyHne29zghL_O+Acmr8+LGG4bCYJpwvOdaQg@mail.gmail.com>
	<CAKbxfG_R2GYs1TfABVpTEFY_zBDD51xguEp5jkAqhkbPyt_h2g@mail.gmail.com>
	<CACyT-3mXz0qQ0KS_BKp06jNeVfJHXr2=T2-SwpzWE-MVj7=S=Q@mail.gmail.com>
	<CAKbxfG_hmhmZzR7sCs1PW-2wF72auiJESD4pPkNNOjt7jenkRA@mail.gmail.com>
	<CACyT-3=iy4G4ZxfG_N3iTgQZZSRLGi2D1RvX7U0SBXpgrAWVyA@mail.gmail.com>
	<CAKbxfG_Sf9zy25CsKFbr5wJt3c_2kNGZs7benpzuqsfcWTqCBg@mail.gmail.com>
	<CACyT-3mwg8GMuDgkGQ+593vsqNne4aVX3BXKjMzBgrBnEQwDyw@mail.gmail.com>
Message-ID: <CAKbxfG-dHEeUv9KWMWP-9tHW1DmB1Rs942s-avghRHQ43PbmBQ@mail.gmail.com>

I think the missing point is that you are terminating point for the
user so the foo+1234 *is* what people use as the user name when
talking to the system.   Whatever you clearly are not interested in
that......

You basically want to modify every channel to fire an event when it
gets an inbound chat message so anybody can subscribe to them.
SWITCH_EVENT_MESSAGE is already used for something you will need
SWITCH_EVENT_RECV_MESSAGE

similar to mod_dptools.c 3260

and put it at sofia_presence.c 2921 inside the else


this really should be wrapped into a profile param to enable it so
it's not firing all these events unless someone really wants it to.
see endless config options being parsed in config_sofia and
reconfig_sofia that set profile flags......


Now tell all your friends at Berkeley to come to ClueCon this year and
we'll see about a student discount!











On Wed, Jul 6, 2011 at 7:50 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
> Let me repeat myself: the changes I am proposing will be backwards
> compatible. Nothing will be broken.
>
> The dialplan abstraction is tangential to this whole discussion. I can
> build a mod foo which runs the dialplan, and have all messages routed
> to it via the + mechanism. This will probably be my initial version of
> the system. I understand the work you put into this, and that your
> decisions were well-reasoned for the use cases you envisioned (e.g.,
> conference calls), but the abstraction isn't working for me. This
> system is just a piece, interoperating with a variety of other
> systems. Attaching a specific module tag, used by just one piece of
> one program, to a username in use throughout the entire system is not
> an elegant solution. This means I have to put a proxy in front of FS,
> adding this + tag to every message coming in.
>
> I'm here and open to any solutions you might have to make the existing
> solution work for me. Fundamentally, I can't just shove a +tag onto
> every incoming message in any elegant way, so far as I know. I think
> that means this entire line of reasoning is just broken for me. Don't
> get me wrong here, I would be really really happy if I didn't have to
> touch sofia to solve this, it's really hard to grok. It's also a lot
> less work for me.
>
> So far, I've proposed one alternate method for doing this. I want
> sofia to generate events and not route messages itself. Then I can
> listen to these perform whatever actions I need to. This may not work,
> and I'm open to implementing any other architectural changes you think
> would enable my use case. You're clearly the guy who knows what to do,
> and your advice is invaluable.
>
> On Wed, Jul 6, 2011 at 5:23 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> you control the user names on your system the prefix+ string is just
>> part of the username people would talk to.
>>
>> for instance ext+ namespace is has auto presence built in so if you
>> add ext+1000 to your list it will auto advertise that it's online.
>>
>> conf+ goes to the conference.
>>
>> conf+888 at server.com will go to FS on server.com and route the chat
>> messages to the mod_conference.
>>
>> so basically you are proposing to try to modify sofia and break the
>> other stuff ?because you do not like the prefix string on the user
>> names but its designed to allow you to cross connect protocols, eg
>> mod_sofia has sip+ registered and mod_dingaling has jingle+ so you can
>> use jabber to pass sip+user at domain.com to your buddy list and have it
>> report accurate presence.
>>
>> Basically you make a new module called mod_foo and register the foo namespace.
>> Then on your clients you subscribe to foo+1000 at server.com for instance.
>> Chat messages to that foo namespace will arrive in your code and you
>> have the option to reply to them completely agnostic of the protocol
>> sip, jabber etc.
>>
>> You seem to be asking for advice but then ignoring it coming from the
>> author of all of the above who spent many man hours solving this
>> problem so it would be abstract. ?so I'm lost for what else to tell
>> you.
>>
>> Maybe we can confirm that you are simply taken aback by the idea of
>> starting all your usernames with foo+ and simply propose to unravel
>> everything in pursuit of removing it?
>>
>>
>>
>>
>>
>> On Wed, Jul 6, 2011 at 6:14 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>> Nope, normal SIP MESSAGE events, no invites.
>>>
>>> I just need an easy place where an app can look up some variables in
>>> the directory, check presence information, and start a call based on
>>> an incoming chat event. I also need a system that interops with
>>> existing SIP chat services. I don't think the +routing does these for
>>> me.
>>>
>>> On Wed, Jul 6, 2011 at 8:46 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> Are we talking about that ridiculous new chat concept in SIP where it
>>>> uses INVITES and dialogs?
>>>>
>>>>
>>>> On Wed, Jul 6, 2011 at 12:56 AM, Kurtis Heimerl
>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>> I think I understand it. Basically, a client can prepend a +code that
>>>>> causes the message to be routed to a specific module. I don't think
>>>>> that fulfills my requirements. This is fine for basic routing, but
>>>>> it's not a very good programmable environment for chat applications,
>>>>> unless I'm missing something, which is always possible.
>>>>>
>>>>> If I am, can you direct me to anything that explains it in a little
>>>>> more depth than your email?
>>>>>
>>>>> On Tue, Jul 5, 2011 at 10:49 PM, Anthony Minessale
>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>> Read my response and my mod conference example again.
>>>>>> You can register a chat callback from any module bound to a particular
>>>>>> namespace.......
>>>>>>
>>>>>> On Jul 6, 2011 12:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>>>>>>> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl
>>>>>>> <kheimerl at cs.berkeley.edu>wrote:
>>>>>>>
>>>>>>>> I don't think so. I set the ESL to log all events (/event all) from
>>>>>>>> the command line and received no events when sofia received SIMPLE
>>>>>>>> messages. Is that a bug? I had assumed it's intentional.
>>>>>>>>
>>>>>>> I'm 99% sure that those messages never make it up into FS from Sofia. I'd
>>>>>>> go
>>>>>>> look into the code but Sofia is really scary and there's no amount of
>>>>>>> Scooby
>>>>>>> Snacks you could give me to convince me to wander into that creepy old
>>>>>>> code.
>>>>>>> :)
>>>>>>> -MC
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-dev mailing list
>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-dev mailing list
>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From kheimerl at cs.berkeley.edu  Thu Jul  7 05:46:33 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Wed, 6 Jul 2011 18:46:33 -0700
Subject: [Freeswitch-dev] Chat Dialplan
In-Reply-To: <CAKbxfG-dHEeUv9KWMWP-9tHW1DmB1Rs942s-avghRHQ43PbmBQ@mail.gmail.com>
References: <CACyT-3ka5PS4mTE1NcjuOSO+SYjPyfugpUQrFBXJTs=k83U6Rg@mail.gmail.com>
	<CACyT-3n19+n83cLP9ovuw7kRtQTVJPbw9Ynf34r9x4vWB9v=QQ@mail.gmail.com>
	<CABH8QX3nj60A24ncWy4L19ifxJAgmdDoKG_8xzceiLAMSmkOgw@mail.gmail.com>
	<CACyT-3=y0-hrajNXJ64KMDJ=c70q48i1Ykju53iqQ_ziLRx3BA@mail.gmail.com>
	<CAKzWOxWFtGYsswjyHne29zghL_O+Acmr8+LGG4bCYJpwvOdaQg@mail.gmail.com>
	<CAKbxfG_R2GYs1TfABVpTEFY_zBDD51xguEp5jkAqhkbPyt_h2g@mail.gmail.com>
	<CACyT-3mXz0qQ0KS_BKp06jNeVfJHXr2=T2-SwpzWE-MVj7=S=Q@mail.gmail.com>
	<CAKbxfG_hmhmZzR7sCs1PW-2wF72auiJESD4pPkNNOjt7jenkRA@mail.gmail.com>
	<CACyT-3=iy4G4ZxfG_N3iTgQZZSRLGi2D1RvX7U0SBXpgrAWVyA@mail.gmail.com>
	<CAKbxfG_Sf9zy25CsKFbr5wJt3c_2kNGZs7benpzuqsfcWTqCBg@mail.gmail.com>
	<CACyT-3mwg8GMuDgkGQ+593vsqNne4aVX3BXKjMzBgrBnEQwDyw@mail.gmail.com>
	<CAKbxfG-dHEeUv9KWMWP-9tHW1DmB1Rs942s-avghRHQ43PbmBQ@mail.gmail.com>
Message-ID: <CACyT-3keJF+1PeKd=nxA+Mhn+TS0LLk1XKzhO_0ag4f9XK18eg@mail.gmail.com>

This particular work is in combination with OpenBTS, and they are
terrible about user names. I basically need as much flexibility as
possible, because the SMSC/BTS/HLR are all likely to tell me exactly
what user names they're willing to accept. Considering it's likely I'm
going to have to work with existing HLRs...  you see the problem.

Thanks for the direction! That will speed this process up
significantly. There are a bunch of us using freeswitch here at
Berkeley. Awaaz.de (http://awaaz.de/) for instance, is helping
hundreds of farmers in rural India right now. I'd love to come out for
cluecon, buy you a beer, and argue in person. We'll see if my advisor
is willing to pay for such a trip!

On Wed, Jul 6, 2011 at 6:21 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I think the missing point is that you are terminating point for the
> user so the foo+1234 *is* what people use as the user name when
> talking to the system. ? Whatever you clearly are not interested in
> that......
>
> You basically want to modify every channel to fire an event when it
> gets an inbound chat message so anybody can subscribe to them.
> SWITCH_EVENT_MESSAGE is already used for something you will need
> SWITCH_EVENT_RECV_MESSAGE
>
> similar to mod_dptools.c 3260
>
> and put it at sofia_presence.c 2921 inside the else
>
>
> this really should be wrapped into a profile param to enable it so
> it's not firing all these events unless someone really wants it to.
> see endless config options being parsed in config_sofia and
> reconfig_sofia that set profile flags......
>
>
> Now tell all your friends at Berkeley to come to ClueCon this year and
> we'll see about a student discount!
>
>
>
>
>
>
>
>
>
>
>
> On Wed, Jul 6, 2011 at 7:50 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>> Let me repeat myself: the changes I am proposing will be backwards
>> compatible. Nothing will be broken.
>>
>> The dialplan abstraction is tangential to this whole discussion. I can
>> build a mod foo which runs the dialplan, and have all messages routed
>> to it via the + mechanism. This will probably be my initial version of
>> the system. I understand the work you put into this, and that your
>> decisions were well-reasoned for the use cases you envisioned (e.g.,
>> conference calls), but the abstraction isn't working for me. This
>> system is just a piece, interoperating with a variety of other
>> systems. Attaching a specific module tag, used by just one piece of
>> one program, to a username in use throughout the entire system is not
>> an elegant solution. This means I have to put a proxy in front of FS,
>> adding this + tag to every message coming in.
>>
>> I'm here and open to any solutions you might have to make the existing
>> solution work for me. Fundamentally, I can't just shove a +tag onto
>> every incoming message in any elegant way, so far as I know. I think
>> that means this entire line of reasoning is just broken for me. Don't
>> get me wrong here, I would be really really happy if I didn't have to
>> touch sofia to solve this, it's really hard to grok. It's also a lot
>> less work for me.
>>
>> So far, I've proposed one alternate method for doing this. I want
>> sofia to generate events and not route messages itself. Then I can
>> listen to these perform whatever actions I need to. This may not work,
>> and I'm open to implementing any other architectural changes you think
>> would enable my use case. You're clearly the guy who knows what to do,
>> and your advice is invaluable.
>>
>> On Wed, Jul 6, 2011 at 5:23 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> you control the user names on your system the prefix+ string is just
>>> part of the username people would talk to.
>>>
>>> for instance ext+ namespace is has auto presence built in so if you
>>> add ext+1000 to your list it will auto advertise that it's online.
>>>
>>> conf+ goes to the conference.
>>>
>>> conf+888 at server.com will go to FS on server.com and route the chat
>>> messages to the mod_conference.
>>>
>>> so basically you are proposing to try to modify sofia and break the
>>> other stuff ?because you do not like the prefix string on the user
>>> names but its designed to allow you to cross connect protocols, eg
>>> mod_sofia has sip+ registered and mod_dingaling has jingle+ so you can
>>> use jabber to pass sip+user at domain.com to your buddy list and have it
>>> report accurate presence.
>>>
>>> Basically you make a new module called mod_foo and register the foo namespace.
>>> Then on your clients you subscribe to foo+1000 at server.com for instance.
>>> Chat messages to that foo namespace will arrive in your code and you
>>> have the option to reply to them completely agnostic of the protocol
>>> sip, jabber etc.
>>>
>>> You seem to be asking for advice but then ignoring it coming from the
>>> author of all of the above who spent many man hours solving this
>>> problem so it would be abstract. ?so I'm lost for what else to tell
>>> you.
>>>
>>> Maybe we can confirm that you are simply taken aback by the idea of
>>> starting all your usernames with foo+ and simply propose to unravel
>>> everything in pursuit of removing it?
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Jul 6, 2011 at 6:14 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>>> Nope, normal SIP MESSAGE events, no invites.
>>>>
>>>> I just need an easy place where an app can look up some variables in
>>>> the directory, check presence information, and start a call based on
>>>> an incoming chat event. I also need a system that interops with
>>>> existing SIP chat services. I don't think the +routing does these for
>>>> me.
>>>>
>>>> On Wed, Jul 6, 2011 at 8:46 AM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>>> Are we talking about that ridiculous new chat concept in SIP where it
>>>>> uses INVITES and dialogs?
>>>>>
>>>>>
>>>>> On Wed, Jul 6, 2011 at 12:56 AM, Kurtis Heimerl
>>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>>> I think I understand it. Basically, a client can prepend a +code that
>>>>>> causes the message to be routed to a specific module. I don't think
>>>>>> that fulfills my requirements. This is fine for basic routing, but
>>>>>> it's not a very good programmable environment for chat applications,
>>>>>> unless I'm missing something, which is always possible.
>>>>>>
>>>>>> If I am, can you direct me to anything that explains it in a little
>>>>>> more depth than your email?
>>>>>>
>>>>>> On Tue, Jul 5, 2011 at 10:49 PM, Anthony Minessale
>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>> Read my response and my mod conference example again.
>>>>>>> You can register a chat callback from any module bound to a particular
>>>>>>> namespace.......
>>>>>>>
>>>>>>> On Jul 6, 2011 12:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>>>>>>>> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl
>>>>>>>> <kheimerl at cs.berkeley.edu>wrote:
>>>>>>>>
>>>>>>>>> I don't think so. I set the ESL to log all events (/event all) from
>>>>>>>>> the command line and received no events when sofia received SIMPLE
>>>>>>>>> messages. Is that a bug? I had assumed it's intentional.
>>>>>>>>>
>>>>>>>> I'm 99% sure that those messages never make it up into FS from Sofia. I'd
>>>>>>>> go
>>>>>>>> look into the code but Sofia is really scary and there's no amount of
>>>>>>>> Scooby
>>>>>>>> Snacks you could give me to convince me to wander into that creepy old
>>>>>>>> code.
>>>>>>>> :)
>>>>>>>> -MC
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-dev mailing list
>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-dev mailing list
>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-dev mailing list
>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From leucaruth at gmail.com  Thu Jul  7 15:35:07 2011
From: leucaruth at gmail.com (Nacho)
Date: Thu, 7 Jul 2011 13:35:07 +0200
Subject: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
In-Reply-To: <35195A50-B87B-4970-BA7E-359849EDE8B2@gmail.com>
References: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>
	<CAKDfjgce=KSUY84+c5LwnqwSR2OrRVSwyM+BgKX=JcqVzK4JpA@mail.gmail.com>
	<CAKbxfG_hNNWf_jC5pT_-9uUpALjuZTBNJE2saSG_K-4-HFQY0g@mail.gmail.com>
	<35195A50-B87B-4970-BA7E-359849EDE8B2@gmail.com>
Message-ID: <CABSXS28dGTtdKqohSUCan+CjcBY9P8dFys9y=XH6LzrpXVpOUA@mail.gmail.com>

Hi again

Im doing as Kristian suggested, at first is an easier solution that making a
module.

I created a plan that redirects calls to a local port where I will do the
testings with the server that will work with that call.

The call is parked and the connection is stablished as I can see with
netcat, so I was thinking about making a program with C or Java that will
become the custom server. The problem is that I don't find info about the
structure of the packages I'll receive from Freeswitch, how to deal with
them or even how can I send orders to freeswitch.

Any suggestions about where I could find some info about it would be kindly
appreciated.

Regards,

Jose.

2011/7/6 Robert-iPhone <rhuddleston at gmail.com>

> I have interest too! Considering mwi / voicemail push for apple
>
> Sent from my iPhone
>
> On Jul 5, 2011, at 5:49 PM, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> > it can be left out of FS code but still part of contrib if an external
> > solution is developed.
> >
> >
> > On Tue, Jul 5, 2011 at 1:20 PM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
> >> Jose,
> >>
> >>  Whether it's C2DM or APNS (Apple) this functionality is best left
> >> out of FreeSWITCH.  FreeSWITCH has plenty of other existing means
> >> (XML_CURL, socket, etc) to drive dynamic call functionality and
> >> interact with other technologies.
> >>
> >>  How do you ask?  An imcoming call to FreeSWITCH could hit static
> >> dialplan and execute a socket connection to some other custom server.
> >> This server could tell FreeSWITCH to do something (play media,
> >> ringback, messages, whatever) while it sends a PUSH (via C2DM or APNS,
> >> for example).
> >>
> >>  One could implement such a socket program easily.  It can be
> >> extended and maintained separately as new push technologies become
> >> available.
> >>
> >> On Mon, Jul 4, 2011 at 7:48 AM, Nacho <leucaruth at gmail.com> wrote:
> >>> Hello all.
> >>> My name is Jose. This is my first post here. I joined the list because
> I was
> >>> wondering about developing a feature for Freeswitch and I would like to
> know
> >>> your opinion.
> >>> The idea is about making a non-persistent client for cell phones that
> works
> >>> with PUSH technology (C2DM and Android for example). These clients
> wouldn't
> >>> be connected to the Freeswitch server at first, but if there is an
> incoming
> >>> call, the Freeswitch server would send a PUSH message to these clients,
> the
> >>> client would process it and, if accepted by the user, the client would
> >>> awake, connect to Freeswitch and then receive the the call invite
> message
> >>> and accept it automatically.
> >>> I don't know if this idea is plausible due to the real time
> restrictions we
> >>> have to face in phone calls. If the delay introduced by C2DM delivery
> is
> >>> high, the waiting time for the caller is probably something that he
> isn't
> >>> willing to accept in order to get his call answered.
> >>> Another question is about the battery life time it would save because
> the
> >>> application woulnd't need to be answering the "still alive" ACKS from
> >>> Freeswitch.
> >>> I'm new in freeswitch developing and still learning about it, but I'm
> really
> >>> interested in this, so any help would be kindly appreciated.
> >>> --
> >>> Aquellos que hablan son esclavos de sus palabras y los que callan
> due?os de
> >>> su silencio.
> >>>
> >>> _______________________________________________
> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>> http://www.cluecon.com 877-7-4ACLUE
> >>>
> >>> FreeSWITCH-dev mailing list
> >>> FreeSWITCH-dev at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >>
> >> --
> >> Kristian Kielhofner
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-dev mailing list
> >> FreeSWITCH-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Aquellos que hablan son esclavos de sus palabras y los que callan due?os de
su silencio.
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From tculjaga at gmail.com  Thu Jul  7 16:50:14 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 7 Jul 2011 14:50:14 +0200
Subject: [Freeswitch-dev] How to Submit patches?
In-Reply-To: <CAKzWOxUYRpWPKa3dFAaz5KU4D5_bxAfC9U1cqQGG3etGzrYnVA@mail.gmail.com>
References: <CAJoKKdX9TSvw+ax=whnMYzB7jh1a5HkA8mjHjCPEw6YGt=wCcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E55E00D4@cooper>
	<CAJoKKdXgbHt56c5BW8duRLyPATyHik4fjN02E6U5GoMExoKxhw@mail.gmail.com>
	<CAO47Hpegj70-Ep-vPDs6SCPFZ7M2VZLUhivC3NfyQMu1eQJNrg@mail.gmail.com>
	<CAKzWOxXxcKs=gkS9OwZVa4KO3JEVf5Sjsu-aW80w=3kpLM2KDQ@mail.gmail.com>
	<CAO47HpdCpp_X1-RASuXbDrsFWruKuq9zFocVcsONgGKS+6N9RA@mail.gmail.com>
	<CAKzWOxUYRpWPKa3dFAaz5KU4D5_bxAfC9U1cqQGG3etGzrYnVA@mail.gmail.com>
Message-ID: <CAJoKKdVV4NPHNYvH_gCg8-W3zE1R82nviEMRGSmPb+-5QbFNtw@mail.gmail.com>

On Wed, Jul 6, 2011 at 2:33 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Jul 5, 2011 at 5:25 PM, jay binks <jaybinks at gmail.com> wrote:
>
>> haha .... I like that one  :)
>>
>

better use this one: http://jira.freeswitch.org/browse/FS-3397
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From kris at kriskinc.com  Thu Jul  7 17:02:32 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 7 Jul 2011 09:02:32 -0400
Subject: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
In-Reply-To: <CABSXS28dGTtdKqohSUCan+CjcBY9P8dFys9y=XH6LzrpXVpOUA@mail.gmail.com>
References: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>
	<CAKDfjgce=KSUY84+c5LwnqwSR2OrRVSwyM+BgKX=JcqVzK4JpA@mail.gmail.com>
	<CAKbxfG_hNNWf_jC5pT_-9uUpALjuZTBNJE2saSG_K-4-HFQY0g@mail.gmail.com>
	<35195A50-B87B-4970-BA7E-359849EDE8B2@gmail.com>
	<CABSXS28dGTtdKqohSUCan+CjcBY9P8dFys9y=XH6LzrpXVpOUA@mail.gmail.com>
Message-ID: <CAKDfjgcg4nvucxtEk-vXNqLdpkU8wmiiHjroVx+bWLQRdwQ16Q@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Event_Socket_Library

On Thu, Jul 7, 2011 at 7:35 AM, Nacho <leucaruth at gmail.com> wrote:
> Hi again
> Im doing as Kristian suggested, at first is an easier solution that making a
> module.
> I created a plan that redirects calls to a local port where I will do the
> testings with the server that will work with that call.
> The call is parked and the connection is stablished as I can see with
> netcat, so I was thinking about making a program with C or Java that will
> become the custom server. The problem is that I don't find info about the
> structure of the packages I'll receive from Freeswitch, how to deal with
> them or even how can I send orders to freeswitch.
> Any suggestions about where I could find some info about it would be kindly
> appreciated.
> Regards,
> Jose.
> 2011/7/6 Robert-iPhone <rhuddleston at gmail.com>
>>
>> I have interest too! Considering mwi / voicemail push for apple
>>
>> Sent from my iPhone
>>
>> On Jul 5, 2011, at 5:49 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>
>> > it can be left out of FS code but still part of contrib if an external
>> > solution is developed.
>> >
>> >
>> > On Tue, Jul 5, 2011 at 1:20 PM, Kristian Kielhofner <kris at kriskinc.com>
>> > wrote:
>> >> Jose,
>> >>
>> >> ?Whether it's C2DM or APNS (Apple) this functionality is best left
>> >> out of FreeSWITCH. ?FreeSWITCH has plenty of other existing means
>> >> (XML_CURL, socket, etc) to drive dynamic call functionality and
>> >> interact with other technologies.
>> >>
>> >> ?How do you ask? ?An imcoming call to FreeSWITCH could hit static
>> >> dialplan and execute a socket connection to some other custom server.
>> >> This server could tell FreeSWITCH to do something (play media,
>> >> ringback, messages, whatever) while it sends a PUSH (via C2DM or APNS,
>> >> for example).
>> >>
>> >> ?One could implement such a socket program easily. ?It can be
>> >> extended and maintained separately as new push technologies become
>> >> available.
>> >>
>> >> On Mon, Jul 4, 2011 at 7:48 AM, Nacho <leucaruth at gmail.com> wrote:
>> >>> Hello all.
>> >>> My name is Jose. This is my first post here. I joined the list because
>> >>> I was
>> >>> wondering about developing a feature for Freeswitch and I would like
>> >>> to know
>> >>> your opinion.
>> >>> The idea is about making a non-persistent client for cell phones that
>> >>> works
>> >>> with PUSH technology (C2DM and Android for example). These clients
>> >>> wouldn't
>> >>> be connected to the Freeswitch server at first, but if there is an
>> >>> incoming
>> >>> call, the Freeswitch server would send a PUSH message to these
>> >>> clients, the
>> >>> client would process it and, if accepted by the user, the client would
>> >>> awake, connect to Freeswitch and then receive the the call invite
>> >>> message
>> >>> and accept it automatically.
>> >>> I don't know if this idea is plausible due to the real time
>> >>> restrictions we
>> >>> have to face in phone calls. If the delay introduced by C2DM delivery
>> >>> is
>> >>> high, the waiting time for the caller is probably something that he
>> >>> isn't
>> >>> willing to accept in order to get his call answered.
>> >>> Another question is about the battery life time it would save because
>> >>> the
>> >>> application woulnd't need to be answering the "still alive" ACKS from
>> >>> Freeswitch.
>> >>> I'm new in freeswitch developing and still learning about it, but I'm
>> >>> really
>> >>> interested in this, so any help would be kindly appreciated.
>> >>> --
>> >>> Aquellos que hablan son esclavos de sus palabras y los que callan
>> >>> due?os de
>> >>> su silencio.
>> >>>
>> >>> _______________________________________________
>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>> http://www.cluecon.com 877-7-4ACLUE
>> >>>
>> >>> FreeSWITCH-dev mailing list
>> >>> FreeSWITCH-dev at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> Kristian Kielhofner
>> >>
>> >> _______________________________________________
>> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> http://www.cluecon.com 877-7-4ACLUE
>> >>
>> >> FreeSWITCH-dev mailing list
>> >> FreeSWITCH-dev at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
>
> --
> Aquellos que hablan son esclavos de sus palabras y los que callan due?os de
> su silencio.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From laurent.habib at voila.fr  Wed Jul  6 15:43:23 2011
From: laurent.habib at voila.fr (laurent.habib at voila.fr)
Date: Wed, 6 Jul 2011 13:43:23 +0200 (CEST)
Subject: [Freeswitch-dev] mod_avmd with wav file or wav buffer
In-Reply-To: <mailman.245.1309952303.24554.freeswitch-dev@lists.freeswitch.org>
Message-ID: <30189351.76071309952603557.JavaMail.www@wwinf4617>

Hello,

Do you have any exemple for detecting beep in wav file with your mod_avmd module ?

(used langage C)

Thanks a lot

Laurent
___________________________________________________________
Les derni?res infos et exclus de stars sont sur Voila people http://people.voila.fr
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From leucaruth at gmail.com  Sun Jul 10 18:12:36 2011
From: leucaruth at gmail.com (Nacho)
Date: Sun, 10 Jul 2011 16:12:36 +0200
Subject: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
In-Reply-To: <CAKDfjgcg4nvucxtEk-vXNqLdpkU8wmiiHjroVx+bWLQRdwQ16Q@mail.gmail.com>
References: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>
	<CAKDfjgce=KSUY84+c5LwnqwSR2OrRVSwyM+BgKX=JcqVzK4JpA@mail.gmail.com>
	<CAKbxfG_hNNWf_jC5pT_-9uUpALjuZTBNJE2saSG_K-4-HFQY0g@mail.gmail.com>
	<35195A50-B87B-4970-BA7E-359849EDE8B2@gmail.com>
	<CABSXS28dGTtdKqohSUCan+CjcBY9P8dFys9y=XH6LzrpXVpOUA@mail.gmail.com>
	<CAKDfjgcg4nvucxtEk-vXNqLdpkU8wmiiHjroVx+bWLQRdwQ16Q@mail.gmail.com>
Message-ID: <CABSXS28zRyDerD4=JmeYzRzb0PKkceDE9M17ot4PYFD2OP=+BQ@mail.gmail.com>

I just noticed short before you answered :).

I have been reading it and I found it very interesting, but I have found a
problem in Java.

The constructor for an Outbound connection requires the File Descriptor (an
Int parameter) of the listening socket to create the ESL object, but as far
as I know, Java doesn't allow to get the int value of a File Descriptor from
a Socket. This would be veary easy to achieve in C, but I have no found any
means to get this to work with Java in Linux (Windows is out of the
question).

I tried to see the source code of the ESL client to see how it uses the int
value, but as I suspected is C code wrapped with SWIG. I have thought about
other ways to work with it, like using the FileDescriptor class in java and
Reflection, but I have not found any way to get the desired int value, so
i'm stuck about what to do right now.

Any ideas? if the constructor requires the int value of the File Descriptor
means that there is a way to get it, doesn't it?

Regards,

Jose

2011/7/7 Kristian Kielhofner <kris at kriskinc.com>

> http://wiki.freeswitch.org/wiki/Event_Socket_Library
>
> On Thu, Jul 7, 2011 at 7:35 AM, Nacho <leucaruth at gmail.com> wrote:
> > Hi again
> > Im doing as Kristian suggested, at first is an easier solution that
> making a
> > module.
> > I created a plan that redirects calls to a local port where I will do the
> > testings with the server that will work with that call.
> > The call is parked and the connection is stablished as I can see with
> > netcat, so I was thinking about making a program with C or Java that will
> > become the custom server. The problem is that I don't find info about the
> > structure of the packages I'll receive from Freeswitch, how to deal with
> > them or even how can I send orders to freeswitch.
> > Any suggestions about where I could find some info about it would be
> kindly
> > appreciated.
> > Regards,
> > Jose.
> > 2011/7/6 Robert-iPhone <rhuddleston at gmail.com>
> >>
> >> I have interest too! Considering mwi / voicemail push for apple
> >>
> >> Sent from my iPhone
> >>
> >> On Jul 5, 2011, at 5:49 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>
> >> > it can be left out of FS code but still part of contrib if an external
> >> > solution is developed.
> >> >
> >> >
> >> > On Tue, Jul 5, 2011 at 1:20 PM, Kristian Kielhofner <
> kris at kriskinc.com>
> >> > wrote:
> >> >> Jose,
> >> >>
> >> >>  Whether it's C2DM or APNS (Apple) this functionality is best left
> >> >> out of FreeSWITCH.  FreeSWITCH has plenty of other existing means
> >> >> (XML_CURL, socket, etc) to drive dynamic call functionality and
> >> >> interact with other technologies.
> >> >>
> >> >>  How do you ask?  An imcoming call to FreeSWITCH could hit static
> >> >> dialplan and execute a socket connection to some other custom server.
> >> >> This server could tell FreeSWITCH to do something (play media,
> >> >> ringback, messages, whatever) while it sends a PUSH (via C2DM or
> APNS,
> >> >> for example).
> >> >>
> >> >>  One could implement such a socket program easily.  It can be
> >> >> extended and maintained separately as new push technologies become
> >> >> available.
> >> >>
> >> >> On Mon, Jul 4, 2011 at 7:48 AM, Nacho <leucaruth at gmail.com> wrote:
> >> >>> Hello all.
> >> >>> My name is Jose. This is my first post here. I joined the list
> because
> >> >>> I was
> >> >>> wondering about developing a feature for Freeswitch and I would like
> >> >>> to know
> >> >>> your opinion.
> >> >>> The idea is about making a non-persistent client for cell phones
> that
> >> >>> works
> >> >>> with PUSH technology (C2DM and Android for example). These clients
> >> >>> wouldn't
> >> >>> be connected to the Freeswitch server at first, but if there is an
> >> >>> incoming
> >> >>> call, the Freeswitch server would send a PUSH message to these
> >> >>> clients, the
> >> >>> client would process it and, if accepted by the user, the client
> would
> >> >>> awake, connect to Freeswitch and then receive the the call invite
> >> >>> message
> >> >>> and accept it automatically.
> >> >>> I don't know if this idea is plausible due to the real time
> >> >>> restrictions we
> >> >>> have to face in phone calls. If the delay introduced by C2DM
> delivery
> >> >>> is
> >> >>> high, the waiting time for the caller is probably something that he
> >> >>> isn't
> >> >>> willing to accept in order to get his call answered.
> >> >>> Another question is about the battery life time it would save
> because
> >> >>> the
> >> >>> application woulnd't need to be answering the "still alive" ACKS
> from
> >> >>> Freeswitch.
> >> >>> I'm new in freeswitch developing and still learning about it, but
> I'm
> >> >>> really
> >> >>> interested in this, so any help would be kindly appreciated.
> >> >>> --
> >> >>> Aquellos que hablan son esclavos de sus palabras y los que callan
> >> >>> due?os de
> >> >>> su silencio.
> >> >>>
> >> >>> _______________________________________________
> >> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> >>> http://www.cluecon.com 877-7-4ACLUE
> >> >>>
> >> >>> FreeSWITCH-dev mailing list
> >> >>> FreeSWITCH-dev at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> >>> http://www.freeswitch.org
> >> >>>
> >> >>>
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> Kristian Kielhofner
> >> >>
> >> >> _______________________________________________
> >> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> >> http://www.cluecon.com 877-7-4ACLUE
> >> >>
> >> >> FreeSWITCH-dev mailing list
> >> >> FreeSWITCH-dev at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org
> >> > googletalk:conf+888 at conference.freeswitch.org
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >
> >> > FreeSWITCH-dev mailing list
> >> > FreeSWITCH-dev at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> > http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-dev mailing list
> >> FreeSWITCH-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Aquellos que hablan son esclavos de sus palabras y los que callan due?os
> de
> > su silencio.
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Aquellos que hablan son esclavos de sus palabras y los que callan due?os de
su silencio.
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From gmaruzz at gmail.com  Tue Jul 12 15:51:05 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 12 Jul 2011 13:51:05 +0200
Subject: [Freeswitch-dev] windows build fails, on XP,
	Visual Studio Exprss 2008: libaprutil - 9 error(s)
Message-ID: <CALXCt0o24XGzT-jtkHn9EjBVjd8RcWQcW1yV3zQXPrEhWhBnkA@mail.gmail.com>

All other code is compiled successfully

------ Build started: Project: libaprutil, Configuration: Debug Win32 ------
Performing Pre-Build Event...
0 File(s) copied
Linking...
   Creating library Debug/libaprutil-1.lib and object Debug/libaprutil-1.exp
apr_xml.obj : error LNK2019: unresolved external symbol
_XML_SetCharacterDataHandler referenced in function _apr_xml_parser_create at 4
apr_xml.obj : error LNK2019: unresolved external symbol
_XML_SetElementHandler referenced in function _apr_xml_parser_create at 4
apr_xml.obj : error LNK2019: unresolved external symbol _XML_SetUserData
referenced in function _apr_xml_parser_create at 4
apr_xml.obj : error LNK2019: unresolved external symbol _XML_ParserCreate
referenced in function _apr_xml_parser_create at 4
apr_xml.obj : error LNK2019: unresolved external symbol _XML_ParserFree
referenced in function _cleanup_parser
apr_xml.obj : error LNK2019: unresolved external symbol _XML_GetErrorCode
referenced in function _do_parse
apr_xml.obj : error LNK2019: unresolved external symbol _XML_Parse
referenced in function _do_parse
apr_xml.obj : error LNK2019: unresolved external symbol _XML_ErrorString
referenced in function _apr_xml_parser_geterror at 12
C:\freeswitch\Debug\libaprutil.dll : fatal error LNK1120: 8 unresolved
externals
Build log was saved at
"file://c:\freeswitch\libs\win32\apr-util\Debug\BuildLog.htm"
libaprutil - 9 error(s), 0 warning(s)
========== Build: 0 succeeded, 1 failed, 1 up-to-date, 0 skipped ==========


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From gmaruzz at gmail.com  Tue Jul 12 15:52:22 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 12 Jul 2011 13:52:22 +0200
Subject: [Freeswitch-dev] windows build fails, on XP,
 Visual Studio Exprss 2008: libaprutil - 9 error(s)
In-Reply-To: <CALXCt0o24XGzT-jtkHn9EjBVjd8RcWQcW1yV3zQXPrEhWhBnkA@mail.gmail.com>
References: <CALXCt0o24XGzT-jtkHn9EjBVjd8RcWQcW1yV3zQXPrEhWhBnkA@mail.gmail.com>
Message-ID: <CALXCt0rUDyd=V+=c3fziKmHHMdvrq+D6UK9R_nU8eJdo9dMRjg@mail.gmail.com>

Build Log

Build started: Project: libaprutil, Configuration: Debug|Win32

Command Lines

Creating temporary file
"c:\freeswitch\libs\win32\apr-util\Debug\BAT00024120802160.bat" with
contents

[
@echo off

if not exist "c:\freeswitch\libs\win32\apr-util\..\..\include\" md
"c:\freeswitch\libs\win32\apr-util\..\..\include\"

if not exist "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_ldap.h"
type "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_ldap.hw"
> "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apr_ldap.h"

if not exist "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu.h"
type "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu.hw"
> "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu.h"

if not exist "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_config.h"
type "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_config.hw"
> "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_config.h"

if not exist "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_select_dbm.h"
type "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_select_dbm.hw"
> "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_select_dbm.h"

if not exist "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_want.h"
type "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_want.hw"
> "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\apu_want.h"

xcopy "c:\freeswitch\libs\win32\apr-util\..\..\apr-util\include\*.h"
"c:\freeswitch\libs\win32\apr-util\..\..\include\" /C /D /Y



if errorlevel 1 goto VCReportError

goto VCEnd

:VCReportError

echo Project : error PRJ0019: A tool returned an error code from
"Performing Pre-Build Event..."

exit 1

:VCEnd
]
Creating command line
"""c:\freeswitch\libs\win32\apr-util\Debug\BAT00024120802160.bat"""

Creating temporary file
"c:\freeswitch\libs\win32\apr-util\Debug\RSP00024220802160.rsp" with
contents
[
/OUT:"C:\freeswitch\Debug\libaprutil.dll" /INCREMENTAL:NO /DLL
/MANIFEST /MANIFESTFILE:"Debug\libaprutil.dll.intermediate.manifest"
/MANIFESTUAC:"level='asInvoker' uiAccess='false'" /DEBUG
/PDB:"Debug/libaprutil-1.pdb" /SUBSYSTEM:WINDOWS /BASE:"0x6EE60000"
/DYNAMICBASE:NO /IMPLIB:"Debug/libaprutil-1.lib" /MACHINE:X86
ws2_32.lib mswsock.lib wldap32.lib kernel32.lib user32.lib gdi32.lib
winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib
oleaut32.lib uuid.lib odbc32.lib odbccp32.lib
"..\apr\debug\libapr-1.lib"

".\Debug\apr_md4.obj"

".\Debug\apr_md5.obj"

".\Debug\apr_sha1.obj"

".\Debug\getuuid.obj"

".\Debug\uuid.obj"

".\Debug\apr_base64.obj"

".\Debug\apr_queue.obj"

".\Debug\xlate.obj"

".\Debug\apr_xml.obj"

".\Debug\libaprutil.res"
]
Creating command line "link.exe
@"c:\freeswitch\libs\win32\apr-util\Debug\RSP00024220802160.rsp"
/NOLOGO /ERRORREPORT:PROMPT"

Output Window

Performing Pre-Build Event...

0 File(s) copied
Linking...
   Creating library Debug/libaprutil-1.lib and object Debug/libaprutil-1.exp
apr_xml.obj : error LNK2019: unresolved external symbol
_XML_SetCharacterDataHandler referenced in function
_apr_xml_parser_create at 4

apr_xml.obj : error LNK2019: unresolved external symbol
_XML_SetElementHandler referenced in function _apr_xml_parser_create at 4
apr_xml.obj : error LNK2019: unresolved external symbol
_XML_SetUserData referenced in function _apr_xml_parser_create at 4

apr_xml.obj : error LNK2019: unresolved external symbol
_XML_ParserCreate referenced in function _apr_xml_parser_create at 4
apr_xml.obj : error LNK2019: unresolved external symbol
_XML_ParserFree referenced in function _cleanup_parser

apr_xml.obj : error LNK2019: unresolved external symbol
_XML_GetErrorCode referenced in function _do_parse
apr_xml.obj : error LNK2019: unresolved external symbol _XML_Parse
referenced in function _do_parse
apr_xml.obj : error LNK2019: unresolved external symbol
_XML_ErrorString referenced in function _apr_xml_parser_geterror at 12

C:\freeswitch\Debug\libaprutil.dll : fatal error LNK1120: 8 unresolved externals

Results

Build log was saved at
"file://c:\freeswitch\libs\win32\apr-util\Debug\BuildLog.htm"


libaprutil - 9 error(s), 0 warning(s)




On Tue, Jul 12, 2011 at 1:51 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> All other code is compiled successfully
>
> ------ Build started: Project: libaprutil, Configuration: Debug Win32
> ------
> Performing Pre-Build Event...
> 0 File(s) copied
> Linking...
>    Creating library Debug/libaprutil-1.lib and object
> Debug/libaprutil-1.exp
> apr_xml.obj : error LNK2019: unresolved external symbol
> _XML_SetCharacterDataHandler referenced in function _apr_xml_parser_create at 4
> apr_xml.obj : error LNK2019: unresolved external symbol
> _XML_SetElementHandler referenced in function _apr_xml_parser_create at 4
> apr_xml.obj : error LNK2019: unresolved external symbol _XML_SetUserData
> referenced in function _apr_xml_parser_create at 4
> apr_xml.obj : error LNK2019: unresolved external symbol _XML_ParserCreate
> referenced in function _apr_xml_parser_create at 4
> apr_xml.obj : error LNK2019: unresolved external symbol _XML_ParserFree
> referenced in function _cleanup_parser
> apr_xml.obj : error LNK2019: unresolved external symbol _XML_GetErrorCode
> referenced in function _do_parse
> apr_xml.obj : error LNK2019: unresolved external symbol _XML_Parse
> referenced in function _do_parse
> apr_xml.obj : error LNK2019: unresolved external symbol _XML_ErrorString
> referenced in function _apr_xml_parser_geterror at 12
> C:\freeswitch\Debug\libaprutil.dll : fatal error LNK1120: 8 unresolved
> externals
> Build log was saved at
> "file://c:\freeswitch\libs\win32\apr-util\Debug\BuildLog.htm"
> libaprutil - 9 error(s), 0 warning(s)
> ========== Build: 0 succeeded, 1 failed, 1 up-to-date, 0 skipped ==========
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From msc at freeswitch.org  Tue Jul 12 21:32:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Jul 2011 10:32:26 -0700
Subject: [Freeswitch-dev] ClueCon Assistance Needed: Sponsoring a VIP
Message-ID: <CAKzWOxWpBjQdzEuRdir43SEqdQCR0m+4Og1jmYjgi0ba893M7A@mail.gmail.com>

Hello all!

If you or your company is in a position to sponsor the travel of a VIP to
ClueCon please contact me off list and I will give you the details. We've
already had one group step up and sponsor Philip Zimmermann (thank you,
Travis!) and we are looking for another one.

Thanks,
Michael
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From debasish.chandra at telemune.net  Tue Jul 12 17:02:18 2011
From: debasish.chandra at telemune.net (Debasish Chandra)
Date: Tue, 12 Jul 2011 18:32:18 +0530
Subject: [Freeswitch-dev] mod_skypopen
Message-ID: <CAE8KqXwnBLGhKUoF4=M1CvkrdGOKcGxhwQednTWDXxh7xfRptw@mail.gmail.com>

Hi,

I am trying to integrate FreeSWITCH with Skype. I have installed skype
client using install.pl under mod_skypopen dir. The skype client is not able
to connect when am running client using start_skype_clients.sh. However it
connect when I open skype101 in skype-clients-symlinks-dir from desktop.

I am using OS: Red Hat Enterprise Linux Server release 5.3 (Tikanga). Below
is the start_skype_clients.sh script
--------------------------------------------------------------------------------------------------------------------------------------------------------------------------
#!/bin/sh
#Unload possible ALSA sound modules that would conflict with our OSS fake
module
rmmod snd_pcm_oss
rmmod snd_mixer_oss
rmmod snd_seq_oss
sleep 1
#Create the inode our fake sound driver will use
mknod /dev/dsp c 14 3
#Load our OSS fake module
insmod /usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko


#start the fake X server on the given port
/usr/bin/Xvfb :101 -ac -nolisten tcp -screen 0 640x480x8 &
sleep 3
# start a Skype client instance that will connect to the X server above, and
will login to the Skype network using the 'username password' you send to it
on stdin.
su root -c "/bin/echo 'username passwd'| DISPLAY=:101
 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype101
--dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype101
--pipelogin &"
sleep 7

exit 0
--------------------------------------------------------------------------------------------------------------------------------------------------------------------------


Can you please help to find out where I am doing wrong?


Best Regards,
Debasish
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From gmaruzz at gmail.com  Wed Jul 13 02:21:26 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 13 Jul 2011 00:21:26 +0200
Subject: [Freeswitch-dev] mod_skypopen
In-Reply-To: <CAE8KqXwnBLGhKUoF4=M1CvkrdGOKcGxhwQednTWDXxh7xfRptw@mail.gmail.com>
References: <CAE8KqXwnBLGhKUoF4=M1CvkrdGOKcGxhwQednTWDXxh7xfRptw@mail.gmail.com>
Message-ID: <CALXCt0pmQd5+vaCHqBmtnOMsfkiPiqOj0Gdh65G1HfWciLwtrA@mail.gmail.com>

You must first use the script to start the skype clients, then start
freeswitch and load the mod-skypopen.

At that point, it works.

Normally you use freeswitch in a server linux installation, not in a
desktop installation.

Anyway, also if you have a desktop installation, you will not see the
skype client on the desktop (because is displayed on a hidden X
screen).

So, all in all, use the script to start the clients, then start fs and
load mod-skypopen. It will work. :)

NB: you don't want to have a skype client running on the desktop. No
skype client on the desktop. Skype clients will run hidden, started by
the script

-giovanni


On 7/12/11, Debasish Chandra <debasish.chandra at telemune.net> wrote:
> Hi,
>
> I am trying to integrate FreeSWITCH with Skype. I have installed skype
> client using install.pl under mod_skypopen dir. The skype client is not able
> to connect when am running client using start_skype_clients.sh. However it
> connect when I open skype101 in skype-clients-symlinks-dir from desktop.
>
> I am using OS: Red Hat Enterprise Linux Server release 5.3 (Tikanga). Below
> is the start_skype_clients.sh script
> --------------------------------------------------------------------------------------------------------------------------------------------------------------------------
> #!/bin/sh
> #Unload possible ALSA sound modules that would conflict with our OSS fake
> module
> rmmod snd_pcm_oss
> rmmod snd_mixer_oss
> rmmod snd_seq_oss
> sleep 1
> #Create the inode our fake sound driver will use
> mknod /dev/dsp c 14 3
> #Load our OSS fake module
> insmod /usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko
>
>
> #start the fake X server on the given port
> /usr/bin/Xvfb :101 -ac -nolisten tcp -screen 0 640x480x8 &
> sleep 3
> # start a Skype client instance that will connect to the X server above, and
> will login to the Skype network using the 'username password' you send to it
> on stdin.
> su root -c "/bin/echo 'username passwd'| DISPLAY=:101
>  /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype101
> --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype101
> --pipelogin &"
> sleep 7
>
> exit 0
> --------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
>
> Can you please help to find out where I am doing wrong?
>
>
> Best Regards,
> Debasish
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From msc at freeswitch.org  Wed Jul 13 20:15:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Jul 2011 09:15:28 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxVE_zSqZKjYXkA50NNjTY7R6Ouv7Hd5-MkaDU5ghPpAWw@mail.gmail.com>

Hello all,

Today's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_13

We have a few things to discuss as a community, plus a cool video to share.
We are especially interested in your thoughts on the subject of FreeSWITCH
"config sets" like what's here:
http://svn.freeswitch.org/svn/configs/

Please be ready to share your config ideas with the group.

Talk to you soon!

-Michael
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From leon at scarlet-internet.nl  Fri Jul 15 20:03:40 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Fri, 15 Jul 2011 18:03:40 +0200
Subject: [Freeswitch-dev] FS-2875 SIP REGISTER Fetch (without Contact header)
Message-ID: <F7A52ECA-104F-4C25-8665-6968943B9263@scarlet-internet.nl>

Hi all,

Can someone please take a look at http://jira.freeswitch.org/browse/FS-2875 ?

Especially the last few comments and last patch which is here: http://jira.freeswitch.org/secure/attachment/14120/sofia_reg_enable_register_query_with_optional_multiple_contacts.patch_4.txt

With this it is possible to fetch bindings by sending a sip register without contact headers. A regular register also responds with all bindings that are in the db now (see traces below).

Thanks,

Leon

---

Traces (I left out headers that are not important for this): 


* UAC does REGISTER fetch (without contact) when there are no bindings in the db yet: 

   REGISTER sip:172.16.44.8 SIP/2.0 
   Via: SIP/2.0/UDP 172.16.42.150:5060;rport;branch=z9hG4bKD33480AB9580F76B 
   From: <sip:1000 at 172.16.44.8>;tag=4041070207 
   To: <sip:1000 at 172.16.44.8> 
   Call-ID: 967655ECA0FEA994 at 192.168.178.1 
   CSeq: 2 REGISTER 

   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP 172.16.42.150:5060;rport=5060;branch=z9hG4bKD33480AB9580F76B 
   From: <sip:1000 at 172.16.44.8>;tag=4041070207 
   To: <sip:1000 at 172.16.44.8>;tag=48QmcmccUc50B 
   Call-ID: 967655ECA0FEA994 at 192.168.178.1 
   CSeq: 2 REGISTER 
   Date: Fri, 15 Jul 2011 10:39:07 GMT 


* UAC does REGISTER (with contact) when there are no bindings in the db yet: 

   REGISTER sip:172.16.44.8 SIP/2.0 
   Via: SIP/2.0/UDP 172.16.42.150:5060;rport;branch=z9hG4bKB8AA233F4F985F7C 
   From: <sip:1000 at 172.16.44.8>;tag=4041070207 
   To: <sip:1000 at 172.16.44.8> 
   Call-ID: 967655ECA0FEA994 at 192.168.178.1 
   CSeq: 3 REGISTER 
   Contact: <sip:1000 at 172.16.42.150;uniq=87E52E228E505C8B8957ADA870829> 
   Expires: 1800 

   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP 172.16.42.150:5060;rport=5060;branch=z9hG4bKB8AA233F4F985F7C 
   From: <sip:1000 at 172.16.44.8>;tag=4041070207 
   To: <sip:1000 at 172.16.44.8>;tag=5HHDeFXFrNUKQ 
   Call-ID: 967655ECA0FEA994 at 192.168.178.1 
   CSeq: 3 REGISTER 
   Contact: <sip:1000 at 172.16.42.150;uniq=87E52E228E505C8B8957ADA870829>;expires=1800 
   Date: Fri, 15 Jul 2011 10:39:07 GMT 


* UAC does REGISTER (with contact) when there is already one binding in the db: 

   REGISTER sip:172.16.44.8 SIP/2.0 
   Via: SIP/2.0/UDP 172.16.42.126:10818;branch=z9hG4bK-d8754z-a7869a477c0f317c-1---d8754z-;rport 
   Contact: <sip:1000 at 172.16.42.126:10818;rinstance=f2aff35495d275bc> 
   To: "1000"<sip:1000 at 172.16.44.8> 
   From: "1000"<sip:1000 at 172.16.44.8>;tag=cf168522 
   Call-ID: MTAyZDJkMTUyN2ViYTE1NmFkNjFiNDc1ZDkwNWQwYTc. 
   CSeq: 2 REGISTER 
   Expires: 3600 

   SIP/2.0 200 OK 
   Via: SIP/2.0/UDP 172.16.42.126:10818;branch=z9hG4bK-d8754z-a7869a477c0f317c-1---d8754z-;rport=10818 
   From: "1000"<sip:1000 at 172.16.44.8>;tag=cf168522 
   To: "1000" <sip:1000 at 172.16.44.8>;tag=ZrQtFUm2rpjtB 
   Call-ID: MTAyZDJkMTUyN2ViYTE1NmFkNjFiNDc1ZDkwNWQwYTc. 
   CSeq: 2 REGISTER 
   Contact: <sip:1000 at 172.16.42.150;uniq=87E52E228E505C8B8957ADA870829>;expires=1787 
   Contact: <sip:1000 at 172.16.42.126:10818;rinstance=f2aff35495d275bc>;expires=3600 
   Date: Fri, 15 Jul 2011 11:08:10 GMT 


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From rhuddleston at gmail.com  Sun Jul 17 23:39:53 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Sun, 17 Jul 2011 15:39:53 -0400
Subject: [Freeswitch-dev] switch_core_sqldb.c
In-Reply-To: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>
References: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>
Message-ID: <FEBDE042-CD20-4BD3-93D9-5827EFA49F83@gmail.com>

Wondering if I should log this to Jira?

Looks like a recent code addition - so not sure protocol here - also never having logged a Jira before.

In switch_core_sqldb.c there are some PRAGMA calls.

I'm finding that in my MYSQL unixODBC environment - these PRAGMA calls are causing syntax errors.

I'm not familiar with the syntax - I've seen equivalents on MS-SQL and Oracle etc - but never seen such on MYSQL.

http://fisheye.freeswitch.org/changelog/freeswitch.git/?showid=11690aff4c4b819e981c4763abff37c33700f5ab&view=fe

Any ideas?

        switch_cache_db_execute_sql(sql_manager.event_db, "PRAGMA synchronous=OFF;", NULL);
        switch_cache_db_execute_sql(sql_manager.event_db, "PRAGMA count_changes=OFF;", NULL);
        switch_cache_db_execute_sql(sql_manager.event_db, "PRAGMA temp_store=MEMORY;", NULL);
        switch_cache_db_execute_sql(sql_manager.event_db, "PRAGMA journal_mode=OFF;", NULL);



From leucaruth at gmail.com  Mon Jul 18 16:20:03 2011
From: leucaruth at gmail.com (Nacho)
Date: Mon, 18 Jul 2011 14:20:03 +0200
Subject: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
In-Reply-To: <CABSXS28jM_ckNi1+2EcVqOwZYtC3EVXO5mLxEj3zeX640kTtmQ@mail.gmail.com>
References: <CABSXS29MaUaW1tQ_DWk50qwbxfpeqvSasUC90DRFYu4gLH4z4g@mail.gmail.com>
	<CAKDfjgce=KSUY84+c5LwnqwSR2OrRVSwyM+BgKX=JcqVzK4JpA@mail.gmail.com>
	<CAKbxfG_hNNWf_jC5pT_-9uUpALjuZTBNJE2saSG_K-4-HFQY0g@mail.gmail.com>
	<35195A50-B87B-4970-BA7E-359849EDE8B2@gmail.com>
	<CABSXS28dGTtdKqohSUCan+CjcBY9P8dFys9y=XH6LzrpXVpOUA@mail.gmail.com>
	<CAKDfjgcg4nvucxtEk-vXNqLdpkU8wmiiHjroVx+bWLQRdwQ16Q@mail.gmail.com>
	<CABSXS28zRyDerD4=JmeYzRzb0PKkceDE9M17ot4PYFD2OP=+BQ@mail.gmail.com>
	<CACj2xxFpJa8ncZ24VySDhdxtQia6bM8WB7AH5RsLjq=B9FfcRw@mail.gmail.com>
	<CABSXS28jM_ckNi1+2EcVqOwZYtC3EVXO5mLxEj3zeX640kTtmQ@mail.gmail.com>
Message-ID: <CABSXS292uQ04utb0J-d=Ec_6Uuq81BSz3KsY88yved9a5yBGNg@mail.gmail.com>

Hi again.

I'm having a little problem with the Java ESL.

I can do some basic things like answer the call, subscribe to events, put
music in the caller... Thought I have som questions about API commands.

The flow right now in my program is this one:


   - I begin listenning at port 8084 for ESL events
   - If I receive an event with the name "channel_data" I begin the
   iteration
   - I answer the call with *sendMsg, addcallcommand("execute")
   and addExecuteAppName("answer");*
   - I subscribe to all events for debuging, I just use *EslMessage response
   = sendSyncSingleLineCommand(channel, "event text all");*
   - I put a tone on the caller  with *sendMsg, addcallcommand("execute")
   and addExecuteAppName("playback");*
   - Now here is a tricky part, I send a petition with HttpURLConnection to
   an APP server, this server is the one that controls the C2DM stuff, he
   answers me when he sends the push and I continue the flow, but still have to
   wait for the legB to connect to the freeswitch server
   - I have to control when the legB gets connected to freeswitch, this is
   something i have to work on, i don't want to be making continuous requests
   to freeswitch to avoid overloading, so right now I assume legB is connected
   by default.
   - It's time to originate the call, so I send *sendMsg,
   addcallcommand("execute"), addExecuteAppName("set")
   and addExecuteAppArg("{ignore_early_media='true'}sofia/internal/1002%10.166.108.139");
   * . *I'm using a default destiny for testing*
   - I have just tested that this also works  *EslMessage response =
   sendSyncSingleLineCommand(channel, "api originate
   {ignore_early_media='true'}sofia/internal/1002%10.166.108.139"); *
   - And finally I bridge the call with SendMsg,
addCallCommand("execute"), addExecuteAppName("bridge"),
addExecuteAppArg("sofia/internal/1002%10.166.108.139")
   and addEventLock();

This works well, but right now my major issue is knowing when the legB gets
connected to freeswitch so I can begin originating the call, but I have some
questions that I would like to know the answer too:


   - The execute messages are for APP so they work like in the dialplan
   <action> field, I saw somewhere that to see the complete list one should
   look in the dialplan.xml, but I haven't found anything about a complete list
   :S
   - Things can be accomplished with API calls too, but I just dont get why
   I have to use "api" before "originate" to make a call, but not needing it
   for the events, it's a little confusing to use API commands like this, is
   there any simpler way to do it?
   - This makes my last question, is there any way to make freeswitch tell
   me when someone registers in the server or do I need to be continuosly
   asking freeswith with an API command if it is connected


Thinks are going well, I have to begin working with the android client next
to be able to process the C2DM messages, right now im using a default voip
client.


Sorry for the long post and thanks for your attention.

Regards

Jose

P.S: Again, thanks to davidv for making possible the use of a Java ESL
client :)


2011/7/11 Nacho <leucaruth at gmail.com>

> Cool, thanks for the tip David, I'm using your library and have it working.
>
> The wiki should advice about the SWIGed library and it's problems with Java
> and outbound connections.
>
> In a few days I will have news about the my progress.
>
> Regards,
>
> Jose
>
> 2011/7/11 david varnes <david.varnes at gmail.com>
>
>> Jose,
>>
>> If you want to use Java and ESL there are other alternatives
>> than the SWIGed library.
>>
>> http://wiki.freeswitch.org/wiki/Java_ESL
>>
>> I maintain the one here http://wiki.freeswitch.org/wiki/Java_ESL_Client
>>
>> I have a few pending changes that I want to commit for a new
>> release, but the existing release is stable.
>>
>> hope this helps
>> davidv
>>
>> On 11 July 2011 00:12, Nacho <leucaruth at gmail.com> wrote:
>> > I just noticed short before you answered :).
>> > I have been reading it and I found it very interesting, but I have found
>> a
>> > problem in Java.
>> > The constructor for an Outbound connection requires the File Descriptor
>> (an
>> > Int parameter) of the listening socket to create the ESL object, but as
>> far
>> > as I know, Java doesn't allow to get the int value of a File Descriptor
>> from
>> > a Socket. This would be veary easy to achieve in C, but I have no found
>> any
>> > means to get this to work with Java in Linux (Windows is out of the
>> > question).
>> > I tried to see the source code of the ESL client to see how it uses the
>> int
>> > value, but as I suspected is C code wrapped with SWIG. I have thought
>> about
>> > other ways to work with it, like using the FileDescriptor class in java
>> and
>> > Reflection, but I have not found any way to get the desired int value,
>> so
>> > i'm stuck about what to do right now.
>> > Any ideas? if the constructor requires the int value of the File
>> Descriptor
>> > means that there is a way to get it, doesn't it?
>> > Regards,
>> > Jose
>> >
>> > 2011/7/7 Kristian Kielhofner <kris at kriskinc.com>
>> >>
>> >> http://wiki.freeswitch.org/wiki/Event_Socket_Library
>> >>
>> >> On Thu, Jul 7, 2011 at 7:35 AM, Nacho <leucaruth at gmail.com> wrote:
>> >> > Hi again
>> >> > Im doing as Kristian suggested, at first is an easier solution that
>> >> > making a
>> >> > module.
>> >> > I created a plan that redirects calls to a local port where I will do
>> >> > the
>> >> > testings with the server that will work with that call.
>> >> > The call is parked and the connection is stablished as I can see with
>> >> > netcat, so I was thinking about making a program with C or Java that
>> >> > will
>> >> > become the custom server. The problem is that I don't find info about
>> >> > the
>> >> > structure of the packages I'll receive from Freeswitch, how to deal
>> with
>> >> > them or even how can I send orders to freeswitch.
>> >> > Any suggestions about where I could find some info about it would be
>> >> > kindly
>> >> > appreciated.
>> >> > Regards,
>> >> > Jose.
>> >> > 2011/7/6 Robert-iPhone <rhuddleston at gmail.com>
>> >> >>
>> >> >> I have interest too! Considering mwi / voicemail push for apple
>> >> >>
>> >> >> Sent from my iPhone
>> >> >>
>> >> >> On Jul 5, 2011, at 5:49 PM, Anthony Minessale
>> >> >> <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> > it can be left out of FS code but still part of contrib if an
>> >> >> > external
>> >> >> > solution is developed.
>> >> >> >
>> >> >> >
>> >> >> > On Tue, Jul 5, 2011 at 1:20 PM, Kristian Kielhofner
>> >> >> > <kris at kriskinc.com>
>> >> >> > wrote:
>> >> >> >> Jose,
>> >> >> >>
>> >> >> >>  Whether it's C2DM or APNS (Apple) this functionality is best
>> left
>> >> >> >> out of FreeSWITCH.  FreeSWITCH has plenty of other existing means
>> >> >> >> (XML_CURL, socket, etc) to drive dynamic call functionality and
>> >> >> >> interact with other technologies.
>> >> >> >>
>> >> >> >>  How do you ask?  An imcoming call to FreeSWITCH could hit static
>> >> >> >> dialplan and execute a socket connection to some other custom
>> >> >> >> server.
>> >> >> >> This server could tell FreeSWITCH to do something (play media,
>> >> >> >> ringback, messages, whatever) while it sends a PUSH (via C2DM or
>> >> >> >> APNS,
>> >> >> >> for example).
>> >> >> >>
>> >> >> >>  One could implement such a socket program easily.  It can be
>> >> >> >> extended and maintained separately as new push technologies
>> become
>> >> >> >> available.
>> >> >> >>
>> >> >> >> On Mon, Jul 4, 2011 at 7:48 AM, Nacho <leucaruth at gmail.com>
>> wrote:
>> >> >> >>> Hello all.
>> >> >> >>> My name is Jose. This is my first post here. I joined the list
>> >> >> >>> because
>> >> >> >>> I was
>> >> >> >>> wondering about developing a feature for Freeswitch and I would
>> >> >> >>> like
>> >> >> >>> to know
>> >> >> >>> your opinion.
>> >> >> >>> The idea is about making a non-persistent client for cell phones
>> >> >> >>> that
>> >> >> >>> works
>> >> >> >>> with PUSH technology (C2DM and Android for example). These
>> clients
>> >> >> >>> wouldn't
>> >> >> >>> be connected to the Freeswitch server at first, but if there is
>> an
>> >> >> >>> incoming
>> >> >> >>> call, the Freeswitch server would send a PUSH message to these
>> >> >> >>> clients, the
>> >> >> >>> client would process it and, if accepted by the user, the client
>> >> >> >>> would
>> >> >> >>> awake, connect to Freeswitch and then receive the the call
>> invite
>> >> >> >>> message
>> >> >> >>> and accept it automatically.
>> >> >> >>> I don't know if this idea is plausible due to the real time
>> >> >> >>> restrictions we
>> >> >> >>> have to face in phone calls. If the delay introduced by C2DM
>> >> >> >>> delivery
>> >> >> >>> is
>> >> >> >>> high, the waiting time for the caller is probably something that
>> he
>> >> >> >>> isn't
>> >> >> >>> willing to accept in order to get his call answered.
>> >> >> >>> Another question is about the battery life time it would save
>> >> >> >>> because
>> >> >> >>> the
>> >> >> >>> application woulnd't need to be answering the "still alive" ACKS
>> >> >> >>> from
>> >> >> >>> Freeswitch.
>> >> >> >>> I'm new in freeswitch developing and still learning about it,
>> but
>> >> >> >>> I'm
>> >> >> >>> really
>> >> >> >>> interested in this, so any help would be kindly appreciated.
>> >> >> >>> --
>> >> >> >>> Aquellos que hablan son esclavos de sus palabras y los que
>> callan
>> >> >> >>> due?os de
>> >> >> >>> su silencio.
>> >> >> >>>
>> >> >> >>> _______________________________________________
>> >> >> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> >> >>> http://www.cluecon.com 877-7-4ACLUE
>> >> >> >>>
>> >> >> >>> FreeSWITCH-dev mailing list
>> >> >> >>> FreeSWITCH-dev at lists.freeswitch.org
>> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> >> >>>
>> >> >> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> >> >>> http://www.freeswitch.org
>> >> >> >>>
>> >> >> >>>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> --
>> >> >> >> Kristian Kielhofner
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> >> >> http://www.cluecon.com 877-7-4ACLUE
>> >> >> >>
>> >> >> >> FreeSWITCH-dev mailing list
>> >> >> >> FreeSWITCH-dev at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> >> >> http://www.freeswitch.org
>> >> >> >>
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > --
>> >> >> > Anthony Minessale II
>> >> >> >
>> >> >> > FreeSWITCH http://www.freeswitch.org/
>> >> >> > ClueCon http://www.cluecon.com/
>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >> >
>> >> >> > AIM: anthm
>> >> >> > MSN:anthony_minessale at hotmail.com
>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >> > IRC: irc.freenode.net #freeswitch
>> >> >> >
>> >> >> > FreeSWITCH Developer Conference
>> >> >> > sip:888 at conference.freeswitch.org
>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> >> > pstn:+19193869900
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> >> > http://www.cluecon.com 877-7-4ACLUE
>> >> >> >
>> >> >> > FreeSWITCH-dev mailing list
>> >> >> > FreeSWITCH-dev at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> >> >
>> >> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> >> > http://www.freeswitch.org
>> >> >>
>> >> >> _______________________________________________
>> >> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> >> http://www.cluecon.com 877-7-4ACLUE
>> >> >>
>> >> >> FreeSWITCH-dev mailing list
>> >> >> FreeSWITCH-dev at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Aquellos que hablan son esclavos de sus palabras y los que callan
>> due?os
>> >> > de
>> >> > su silencio.
>> >> >
>> >> > _______________________________________________
>> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> > http://www.cluecon.com 877-7-4ACLUE
>> >> >
>> >> > FreeSWITCH-dev mailing list
>> >> > FreeSWITCH-dev at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Kristian Kielhofner
>> >>
>> >> _______________________________________________
>> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> http://www.cluecon.com 877-7-4ACLUE
>> >>
>> >> FreeSWITCH-dev mailing list
>> >> FreeSWITCH-dev at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Aquellos que hablan son esclavos de sus palabras y los que callan due?os
>> de
>> > su silencio.
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> david varnes
>>
>> e: david.varnes at gmail.com
>> p: +61 404 925 633
>>
>
>
>
> --
> Aquellos que hablan son esclavos de sus palabras y los que callan due?os de
> su silencio.
>



-- 
Aquellos que hablan son esclavos de sus palabras y los que callan due?os de
su silencio.
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From david.varnes at gmail.com  Mon Jul 18 18:08:57 2011
From: david.varnes at gmail.com (david varnes)
Date: Tue, 19 Jul 2011 00:08:57 +1000
Subject: [Freeswitch-dev] bounty request - sofia to not route MESSAGE [was]
	Chat Dialplan
Message-ID: <CACj2xxHA1cROMwAMqcfWPT8YcHYeWG2vNANHuv4UdFidBB+3ow@mail.gmail.com>

I raised a bounty jira http://jira.freeswitch.org/browse/FS-3408 which
is a request to modify the native sofia treatment of incoming MESSAGE,
and generate some events to allow alternate handling. I also added
it to the bottom of http://wiki.freeswitch.org/wiki/Bounty

I was just reading back through the list and saw this discussion,
it seems pretty much what Kurtis was suggesting as well ?

Given the rest of the previous thread, is this request (FS-3408) likely to
get any traction ?  Is there anything else I should/could do to get a
yes/no answer.  The reason I ask is that I have a client that would
like to use this kind of feature, and will try and do the modification
themselves if the change does not fit with FS dev direction. I think
it would be much better if the change was available in FS though.
What amount of work would it be ?

thanks for any feedback  :-)

davidv


On 7 July 2011 10:50, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
> Let me repeat myself: the changes I am proposing will be backwards
> compatible. Nothing will be broken.
>
> The dialplan abstraction is tangential to this whole discussion. I can
> build a mod foo which runs the dialplan, and have all messages routed
> to it via the + mechanism. This will probably be my initial version of
> the system. I understand the work you put into this, and that your
> decisions were well-reasoned for the use cases you envisioned (e.g.,
> conference calls), but the abstraction isn't working for me. This
> system is just a piece, interoperating with a variety of other
> systems. Attaching a specific module tag, used by just one piece of
> one program, to a username in use throughout the entire system is not
> an elegant solution. This means I have to put a proxy in front of FS,
> adding this + tag to every message coming in.
>
> I'm here and open to any solutions you might have to make the existing
> solution work for me. Fundamentally, I can't just shove a +tag onto
> every incoming message in any elegant way, so far as I know. I think
> that means this entire line of reasoning is just broken for me. Don't
> get me wrong here, I would be really really happy if I didn't have to
> touch sofia to solve this, it's really hard to grok. It's also a lot
> less work for me.
>
> So far, I've proposed one alternate method for doing this. I want
> sofia to generate events and not route messages itself. Then I can
> listen to these perform whatever actions I need to. This may not work,
> and I'm open to implementing any other architectural changes you think
> would enable my use case. You're clearly the guy who knows what to do,
> and your advice is invaluable.
>
> On Wed, Jul 6, 2011 at 5:23 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> you control the user names on your system the prefix+ string is just
>> part of the username people would talk to.
>>
>> for instance ext+ namespace is has auto presence built in so if you
>> add ext+1000 to your list it will auto advertise that it's online.
>>
>> conf+ goes to the conference.
>>
>> conf+888 at server.com will go to FS on server.com and route the chat
>> messages to the mod_conference.
>>
>> so basically you are proposing to try to modify sofia and break the
>> other stuff ?because you do not like the prefix string on the user
>> names but its designed to allow you to cross connect protocols, eg
>> mod_sofia has sip+ registered and mod_dingaling has jingle+ so you can
>> use jabber to pass sip+user at domain.com to your buddy list and have it
>> report accurate presence.
>>
>> Basically you make a new module called mod_foo and register the foo namespace.
>> Then on your clients you subscribe to foo+1000 at server.com for instance.
>> Chat messages to that foo namespace will arrive in your code and you
>> have the option to reply to them completely agnostic of the protocol
>> sip, jabber etc.
>>
>> You seem to be asking for advice but then ignoring it coming from the
>> author of all of the above who spent many man hours solving this
>> problem so it would be abstract. ?so I'm lost for what else to tell
>> you.
>>
>> Maybe we can confirm that you are simply taken aback by the idea of
>> starting all your usernames with foo+ and simply propose to unravel
>> everything in pursuit of removing it?
>>
>>
>>
>>
>>
>> On Wed, Jul 6, 2011 at 6:14 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>> Nope, normal SIP MESSAGE events, no invites.
>>>
>>> I just need an easy place where an app can look up some variables in
>>> the directory, check presence information, and start a call based on
>>> an incoming chat event. I also need a system that interops with
>>> existing SIP chat services. I don't think the +routing does these for
>>> me.
>>>
>>> On Wed, Jul 6, 2011 at 8:46 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> Are we talking about that ridiculous new chat concept in SIP where it
>>>> uses INVITES and dialogs?
>>>>
>>>>
>>>> On Wed, Jul 6, 2011 at 12:56 AM, Kurtis Heimerl
>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>> I think I understand it. Basically, a client can prepend a +code that
>>>>> causes the message to be routed to a specific module. I don't think
>>>>> that fulfills my requirements. This is fine for basic routing, but
>>>>> it's not a very good programmable environment for chat applications,
>>>>> unless I'm missing something, which is always possible.
>>>>>
>>>>> If I am, can you direct me to anything that explains it in a little
>>>>> more depth than your email?
>>>>>
>>>>> On Tue, Jul 5, 2011 at 10:49 PM, Anthony Minessale
>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>> Read my response and my mod conference example again.
>>>>>> You can register a chat callback from any module bound to a particular
>>>>>> namespace.......
>>>>>>
>>>>>> On Jul 6, 2011 12:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>>>>>>> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl
>>>>>>> <kheimerl at cs.berkeley.edu>wrote:
>>>>>>>
>>>>>>>> I don't think so. I set the ESL to log all events (/event all) from
>>>>>>>> the command line and received no events when sofia received SIMPLE
>>>>>>>> messages. Is that a bug? I had assumed it's intentional.
>>>>>>>>
>>>>>>> I'm 99% sure that those messages never make it up into FS from Sofia. I'd
>>>>>>> go
>>>>>>> look into the code but Sofia is really scary and there's no amount of
>>>>>>> Scooby
>>>>>>> Snacks you could give me to convince me to wander into that creepy old
>>>>>>> code.
>>>>>>> :)
>>>>>>> -MC
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-dev mailing list
>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-dev mailing list
>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>
>> --
>> Anthony Minessale II
>>

-- 
david varnes

e: david.varnes at gmail.com


From kheimerl at cs.berkeley.edu  Tue Jul 19 01:51:07 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 18 Jul 2011 14:51:07 -0700
Subject: [Freeswitch-dev] bounty request - sofia to not route MESSAGE
 [was] Chat Dialplan
In-Reply-To: <CACj2xxHA1cROMwAMqcfWPT8YcHYeWG2vNANHuv4UdFidBB+3ow@mail.gmail.com>
References: <CACj2xxHA1cROMwAMqcfWPT8YcHYeWG2vNANHuv4UdFidBB+3ow@mail.gmail.com>
Message-ID: <CACyT-3mvRW93=mikeNhh+jM2pqF=LR6_9Q0L9nweM2JOXGu=+Q@mail.gmail.com>

I can tell you that I plan on implementing the change, though it's not
on my critical path. I'm in the midst of getting the text dialplan
part of my work going, it's the item afterwards.

It doesn't look like a complicated change and Anthony was kind enough
to direct me to where the code needs to go. I don't expect a lot of
troubles, but we academics are notoriously unreliable.

Whatever happens, I'd like to be updated as to what you folks are
doing, just in case I'm beaten to the punch. I'll return the favor, of
course.

On Mon, Jul 18, 2011 at 7:08 AM, david varnes <david.varnes at gmail.com> wrote:
> I raised a bounty jira http://jira.freeswitch.org/browse/FS-3408 which
> is a request to modify the native sofia treatment of incoming MESSAGE,
> and generate some events to allow alternate handling. I also added
> it to the bottom of http://wiki.freeswitch.org/wiki/Bounty
>
> I was just reading back through the list and saw this discussion,
> it seems pretty much what Kurtis was suggesting as well ?
>
> Given the rest of the previous thread, is this request (FS-3408) likely to
> get any traction ? ?Is there anything else I should/could do to get a
> yes/no answer. ?The reason I ask is that I have a client that would
> like to use this kind of feature, and will try and do the modification
> themselves if the change does not fit with FS dev direction. I think
> it would be much better if the change was available in FS though.
> What amount of work would it be ?
>
> thanks for any feedback ?:-)
>
> davidv
>
>
> On 7 July 2011 10:50, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>> Let me repeat myself: the changes I am proposing will be backwards
>> compatible. Nothing will be broken.
>>
>> The dialplan abstraction is tangential to this whole discussion. I can
>> build a mod foo which runs the dialplan, and have all messages routed
>> to it via the + mechanism. This will probably be my initial version of
>> the system. I understand the work you put into this, and that your
>> decisions were well-reasoned for the use cases you envisioned (e.g.,
>> conference calls), but the abstraction isn't working for me. This
>> system is just a piece, interoperating with a variety of other
>> systems. Attaching a specific module tag, used by just one piece of
>> one program, to a username in use throughout the entire system is not
>> an elegant solution. This means I have to put a proxy in front of FS,
>> adding this + tag to every message coming in.
>>
>> I'm here and open to any solutions you might have to make the existing
>> solution work for me. Fundamentally, I can't just shove a +tag onto
>> every incoming message in any elegant way, so far as I know. I think
>> that means this entire line of reasoning is just broken for me. Don't
>> get me wrong here, I would be really really happy if I didn't have to
>> touch sofia to solve this, it's really hard to grok. It's also a lot
>> less work for me.
>>
>> So far, I've proposed one alternate method for doing this. I want
>> sofia to generate events and not route messages itself. Then I can
>> listen to these perform whatever actions I need to. This may not work,
>> and I'm open to implementing any other architectural changes you think
>> would enable my use case. You're clearly the guy who knows what to do,
>> and your advice is invaluable.
>>
>> On Wed, Jul 6, 2011 at 5:23 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> you control the user names on your system the prefix+ string is just
>>> part of the username people would talk to.
>>>
>>> for instance ext+ namespace is has auto presence built in so if you
>>> add ext+1000 to your list it will auto advertise that it's online.
>>>
>>> conf+ goes to the conference.
>>>
>>> conf+888 at server.com will go to FS on server.com and route the chat
>>> messages to the mod_conference.
>>>
>>> so basically you are proposing to try to modify sofia and break the
>>> other stuff ?because you do not like the prefix string on the user
>>> names but its designed to allow you to cross connect protocols, eg
>>> mod_sofia has sip+ registered and mod_dingaling has jingle+ so you can
>>> use jabber to pass sip+user at domain.com to your buddy list and have it
>>> report accurate presence.
>>>
>>> Basically you make a new module called mod_foo and register the foo namespace.
>>> Then on your clients you subscribe to foo+1000 at server.com for instance.
>>> Chat messages to that foo namespace will arrive in your code and you
>>> have the option to reply to them completely agnostic of the protocol
>>> sip, jabber etc.
>>>
>>> You seem to be asking for advice but then ignoring it coming from the
>>> author of all of the above who spent many man hours solving this
>>> problem so it would be abstract. ?so I'm lost for what else to tell
>>> you.
>>>
>>> Maybe we can confirm that you are simply taken aback by the idea of
>>> starting all your usernames with foo+ and simply propose to unravel
>>> everything in pursuit of removing it?
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Jul 6, 2011 at 6:14 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>>> Nope, normal SIP MESSAGE events, no invites.
>>>>
>>>> I just need an easy place where an app can look up some variables in
>>>> the directory, check presence information, and start a call based on
>>>> an incoming chat event. I also need a system that interops with
>>>> existing SIP chat services. I don't think the +routing does these for
>>>> me.
>>>>
>>>> On Wed, Jul 6, 2011 at 8:46 AM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>>> Are we talking about that ridiculous new chat concept in SIP where it
>>>>> uses INVITES and dialogs?
>>>>>
>>>>>
>>>>> On Wed, Jul 6, 2011 at 12:56 AM, Kurtis Heimerl
>>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>>> I think I understand it. Basically, a client can prepend a +code that
>>>>>> causes the message to be routed to a specific module. I don't think
>>>>>> that fulfills my requirements. This is fine for basic routing, but
>>>>>> it's not a very good programmable environment for chat applications,
>>>>>> unless I'm missing something, which is always possible.
>>>>>>
>>>>>> If I am, can you direct me to anything that explains it in a little
>>>>>> more depth than your email?
>>>>>>
>>>>>> On Tue, Jul 5, 2011 at 10:49 PM, Anthony Minessale
>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>> Read my response and my mod conference example again.
>>>>>>> You can register a chat callback from any module bound to a particular
>>>>>>> namespace.......
>>>>>>>
>>>>>>> On Jul 6, 2011 12:35 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>>>>>>>> On Tue, Jul 5, 2011 at 10:20 PM, Kurtis Heimerl
>>>>>>>> <kheimerl at cs.berkeley.edu>wrote:
>>>>>>>>
>>>>>>>>> I don't think so. I set the ESL to log all events (/event all) from
>>>>>>>>> the command line and received no events when sofia received SIMPLE
>>>>>>>>> messages. Is that a bug? I had assumed it's intentional.
>>>>>>>>>
>>>>>>>> I'm 99% sure that those messages never make it up into FS from Sofia. I'd
>>>>>>>> go
>>>>>>>> look into the code but Sofia is really scary and there's no amount of
>>>>>>>> Scooby
>>>>>>>> Snacks you could give me to convince me to wander into that creepy old
>>>>>>>> code.
>>>>>>>> :)
>>>>>>>> -MC
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-dev mailing list
>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-dev mailing list
>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-dev mailing list
>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>
> --
> david varnes
>
> e: david.varnes at gmail.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From kheimerl at cs.berkeley.edu  Tue Jul 19 04:39:43 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 18 Jul 2011 17:39:43 -0700
Subject: [Freeswitch-dev] Associating a Dialplan with a Session
Message-ID: <CACyT-3nY5Nvf5OpVc2N0hh33ttvD2=UKHS2C=N4wE7+aZyU2Bw@mail.gmail.com>

Hello FS-Dev!

I'm trying to figure out how to get associate a dialplan with a newly
created Session. I create the new session when I receive an chat
message. Here's the current state of the code:
http://pastebin.freeswitch.org/16844

There are (undoubtedly) many things wrong with this code...  but I'm
focused on getting it to route the SMS through the dialplan right now.

It fails in the following way:

...
2011-07-18 15:05:49.315430 [DEBUG] mod_smqueue.c:39 laaa CHANNEL ROUTING
2011-07-18 15:05:49.315430 [DEBUG] switch_core_state_machine.c:77 laaa
Standard ROUTING
2011-07-18 15:05:49.315430 [INFO] switch_core_state_machine.c:142 No
Route, Aborting
2011-07-18 15:05:49.315430 [DEBUG] switch_channel.c:2641 (laaa)
Callstate Change RINGING -> HANGUP
2011-07-18 15:05:49.315430 [NOTICE] switch_core_state_machine.c:143
Hangup laaa [CS_ROUTING] [NO_ROUTE_DESTINATION]
2011-07-18 15:05:49.315430 [DEBUG] switch_channel.c:2657 Send signal laaa [KILL]
...

The "No Route, Aborting" debug message SEEMS to indicate either that
the extension isn't found, or that the dialplan isn't found. I think
I've confirmed it's the dialplan by passing NULL instead of
"globals.dialplan" in my own code and getting the same result. This is
also confirmed by the line:
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "KURTIS %p\n",
(void*) switch_loadable_module_get_dialplan_interface("default"));
printing "(nil)" instead of returning a valid pointer.

I assume there must be some other naming mechanism for dialplans that
I'm not using correctly. I pass "default" right now (as I saw in
mod_dingaling.c) but that's not working.

Any suggestions would be appreciated!


From kheimerl at cs.berkeley.edu  Tue Jul 19 05:03:09 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 18 Jul 2011 18:03:09 -0700
Subject: [Freeswitch-dev] Associating a Dialplan with a Session
In-Reply-To: <CACyT-3nY5Nvf5OpVc2N0hh33ttvD2=UKHS2C=N4wE7+aZyU2Bw@mail.gmail.com>
References: <CACyT-3nY5Nvf5OpVc2N0hh33ttvD2=UKHS2C=N4wE7+aZyU2Bw@mail.gmail.com>
Message-ID: <CACyT-3=nymsxRy5+S4FErVw4=Y_fpqhHS_cynTgybG2zAnt6fg@mail.gmail.com>

Oh I feel sheepish. Writing this email has resolved the issue for me:

the dialplan field is the MODULE type (e.g., "XML")

while context is the actual file to be loaded (e.g., "default")

It's now routing and I hope google picks this message up and steers
confused folks like myself away from future public mailing lists.

On Mon, Jul 18, 2011 at 5:39 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> Hello FS-Dev!
>
> I'm trying to figure out how to get associate a dialplan with a newly
> created Session. I create the new session when I receive an chat
> message. Here's the current state of the code:
> http://pastebin.freeswitch.org/16844
>
> There are (undoubtedly) many things wrong with this code... ?but I'm
> focused on getting it to route the SMS through the dialplan right now.
>
> It fails in the following way:
>
> ...
> 2011-07-18 15:05:49.315430 [DEBUG] mod_smqueue.c:39 laaa CHANNEL ROUTING
> 2011-07-18 15:05:49.315430 [DEBUG] switch_core_state_machine.c:77 laaa
> Standard ROUTING
> 2011-07-18 15:05:49.315430 [INFO] switch_core_state_machine.c:142 No
> Route, Aborting
> 2011-07-18 15:05:49.315430 [DEBUG] switch_channel.c:2641 (laaa)
> Callstate Change RINGING -> HANGUP
> 2011-07-18 15:05:49.315430 [NOTICE] switch_core_state_machine.c:143
> Hangup laaa [CS_ROUTING] [NO_ROUTE_DESTINATION]
> 2011-07-18 15:05:49.315430 [DEBUG] switch_channel.c:2657 Send signal laaa [KILL]
> ...
>
> The "No Route, Aborting" debug message SEEMS to indicate either that
> the extension isn't found, or that the dialplan isn't found. I think
> I've confirmed it's the dialplan by passing NULL instead of
> "globals.dialplan" in my own code and getting the same result. This is
> also confirmed by the line:
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "KURTIS %p\n",
> (void*) switch_loadable_module_get_dialplan_interface("default"));
> printing "(nil)" instead of returning a valid pointer.
>
> I assume there must be some other naming mechanism for dialplans that
> I'm not using correctly. I pass "default" right now (as I saw in
> mod_dingaling.c) but that's not working.
>
> Any suggestions would be appreciated!
>


From jaybinks at gmail.com  Tue Jul 19 11:55:23 2011
From: jaybinks at gmail.com (jay binks)
Date: Tue, 19 Jul 2011 17:55:23 +1000
Subject: [Freeswitch-dev] Urgent Feature Request - RTCP report blocks
Message-ID: <CAO47HpcdY-PSitfd_2njqJNa8OH6aOMP06_oqAvveqZugyFf-A@mail.gmail.com>

I NEED to get Freeswitch generating RTCP Report Blocks
( with all associated data - packet_loss, jitter etc )

Im happy to consider a bounty on this requirement as I
need this done as quickly as humanly possible.

in preference to only filling my request,
a complete RTCP implementation would be great however currently
I simply need RTCP packet generation with Packet Loss Jitter and RTT
information.

we need this is to fulfill an interop test we are trying to get done, so as
you can imagine
time is of the essence.

please contact me if you wish to work on this, to be eligible for bounty
payment
the code will need to be accepted into the FS Git tree and pass my testing.
( however I would like to test asap, even before inclusion in git )

I would like your guidance, as to what bounty you think this would require.
please contact me for bounty approval.
this email does not constitute acceptance of any bounty / work.

Sincerely

Jay Binks
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From msc at freeswitch.org  Wed Jul 20 19:32:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jul 2011 08:32:56 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxWz-tQxvHVvaTRv5B8TKxGdj+5DEvPgo_1AQ6KsEbku9Q@mail.gmail.com>

Hello all!

Today's agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_20

We have a special guest: Alfred from the baresip project is coming in to
talk about his cool new SIP client! Come join us today at 1PM eastern/10AM
pacific/1700 UTC.

-Michael
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From msc at freeswitch.org  Thu Jul 21 00:31:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jul 2011 13:31:21 -0700
Subject: [Freeswitch-dev] ClueCon 2011 - Sharing rooms
Message-ID: <CAKzWOxUmDZyqSmTgummnScQmo5WaiVEhmK2eX3SC=X4MHKdCGg@mail.gmail.com>

Hello all!

It seems that the Sofitel is getting booked up rapidly and it's becoming
difficult to find rooms. If you are looking for someone with whom you may
share a room, or you have a room that you would like to share, please email
me off list.

Thanks,
Michael
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From peter.olsson at visionutveckling.se  Sat Jul 23 17:58:37 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 23 Jul 2011 15:58:37 +0200
Subject: [Freeswitch-dev] Execution of channel functions and thread safety
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F88@cooper>

Hello all developers!

I'm in the middle of the development of my own FreeSWITCH module, written in C. Since I've been following the project for quite some time now, it was pretty easy to get things going :) And most of the Core API is straight forward to use - thanks for that all FS developers!

I have a minor question of some of the channel functions in FreeSWITCH, and if it's safe to call them from different threads (outside the channel's own thread).

For instance; I have a dialplan app registered, and when the call arrives there I do some minor parsing of the call, and after that I park the call (using switch_ivr_park()).

Now to my question, I understand that it's probably not safe to call switch_ivr_play_file() from another thread at this time, since the park app itself will then try to handle RTP at the same time as switch_ivr_play_file() (especially if park app sends silence frames) - so instead I queue this to the "playback" app on the channel, similar to how it's done in mod_event_socket. But, what if I call switch_channel_answer(), switch_channel_pre_answer() etc, is that safe to call from another thread, or should I queue these messages to the channel instead - so they are executed within the channel's own thread?

My first thought was that these were supposed to be executed by the channel's thread only, but after looking at some of the uuid_-commands in mod_commands I'm not so sure anymore. Since there are quite a few commands messing with the channel's states etc, and that would be executed from another thread indeed (for instance hold/unhold, kill etc).

Right now I'm calling switch_channel_answer() from another thread, and it works great, however, I want to be sure I do it right, so this won't cause any problems for me later on in the project.

Regards,

Peter Olsson

PS! See you at ClueCon!


From kheimerl at cs.berkeley.edu  Sun Jul 24 02:36:35 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sat, 23 Jul 2011 15:36:35 -0700
Subject: [Freeswitch-dev] Mod_commands, originate, and existing sessions
Message-ID: <CACyT-3==N35bVTA1V3Qv0S84NQ+hvn4GaJe75hnhCf5jdwRrhw@mail.gmail.com>

Hello freeswitch-dev!

I've got my chat dialplan working, and set about creating a simple
"callback" dialplan. Basically, a chat comes in at a number (e.g.,
5000) and the dialplan immediately originates a call between that user
and that phone number. This didn't work, as the originate command (in
mod_commands) does not allow it to operate when an existing session is
active. I've modified it to allow this, and it works great. Callbacks
happen, a new session is created, and I see no crashing or major
issues.

Line ~3400 in mod_command.c

	//kurtis
	//if (session || zstr(cmd)) {
	if (zstr(cmd)) {
		stream->write_function(stream, "-USAGE %s\n", ORIGINATE_SYNTAX);
		return SWITCH_STATUS_SUCCESS;
	}

I'd like to have as few distinct blocks of code from vanilla FS as
possible, so I was thinking of pushing this change upstream. Would any
of the core developers tell me why this particular solution is a bad
idea, and why originate disallowed operation in an existing session?
If it's benign, I'll submit a patch. If not, I'll try to figure out
another way to do this.

Thanks!


From debasish.chandra at telemune.net  Wed Jul 13 15:21:46 2011
From: debasish.chandra at telemune.net (Debasish Chandra)
Date: Wed, 13 Jul 2011 16:51:46 +0530
Subject: [Freeswitch-dev] mod_skypopen
In-Reply-To: <CALXCt0pmQd5+vaCHqBmtnOMsfkiPiqOj0Gdh65G1HfWciLwtrA@mail.gmail.com>
References: <CAE8KqXwnBLGhKUoF4=M1CvkrdGOKcGxhwQednTWDXxh7xfRptw@mail.gmail.com>
	<CALXCt0pmQd5+vaCHqBmtnOMsfkiPiqOj0Gdh65G1HfWciLwtrA@mail.gmail.com>
Message-ID: <CAE8KqXwZtv4hF=qVTdOCym570hq+xKksdobzFZ6OkJMSqDnAfw@mail.gmail.com>

Hi Giovanni,

Thanks a lot for your reply. I was using old freeswitch snapshot. I have
installed freeswitch from latest Git, and now mod_skypopen is working fine.

Best Regards,
Debasish

On Wed, Jul 13, 2011 at 3:51 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> You must first use the script to start the skype clients, then start
> freeswitch and load the mod-skypopen.
>
> At that point, it works.
>
> Normally you use freeswitch in a server linux installation, not in a
> desktop installation.
>
> Anyway, also if you have a desktop installation, you will not see the
> skype client on the desktop (because is displayed on a hidden X
> screen).
>
> So, all in all, use the script to start the clients, then start fs and
> load mod-skypopen. It will work. :)
>
> NB: you don't want to have a skype client running on the desktop. No
> skype client on the desktop. Skype clients will run hidden, started by
> the script
>
> -giovanni
>
>
> On 7/12/11, Debasish Chandra <debasish.chandra at telemune.net> wrote:
> > Hi,
> >
> > I am trying to integrate FreeSWITCH with Skype. I have installed skype
> > client using install.pl under mod_skypopen dir. The skype client is not
> able
> > to connect when am running client using start_skype_clients.sh. However
> it
> > connect when I open skype101 in skype-clients-symlinks-dir from desktop.
> >
> > I am using OS: Red Hat Enterprise Linux Server release 5.3 (Tikanga).
> Below
> > is the start_skype_clients.sh script
> >
> --------------------------------------------------------------------------------------------------------------------------------------------------------------------------
> > #!/bin/sh
> > #Unload possible ALSA sound modules that would conflict with our OSS fake
> > module
> > rmmod snd_pcm_oss
> > rmmod snd_mixer_oss
> > rmmod snd_seq_oss
> > sleep 1
> > #Create the inode our fake sound driver will use
> > mknod /dev/dsp c 14 3
> > #Load our OSS fake module
> > insmod
> /usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko
> >
> >
> > #start the fake X server on the given port
> > /usr/bin/Xvfb :101 -ac -nolisten tcp -screen 0 640x480x8 &
> > sleep 3
> > # start a Skype client instance that will connect to the X server above,
> and
> > will login to the Skype network using the 'username password' you send to
> it
> > on stdin.
> > su root -c "/bin/echo 'username passwd'| DISPLAY=:101
> >  /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype101
> >
> --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype101
> > --pipelogin &"
> > sleep 7
> >
> > exit 0
> >
> --------------------------------------------------------------------------------------------------------------------------------------------------------------------------
> >
> >
> > Can you please help to find out where I am doing wrong?
> >
> >
> > Best Regards,
> > Debasish
> >
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From dimskraft at gmail.com  Tue Jul 19 15:40:51 2011
From: dimskraft at gmail.com (Dmitry Kravchenko)
Date: Tue, 19 Jul 2011 15:40:51 +0400
Subject: [Freeswitch-dev] Bug in
	http://fisheye.freeswitch.org/browse/freeswitch.git/clients/flex/freeswitch.html
Message-ID: <CADDV4gwSKC8r7-tBCUYN6vwYCUzVET5S+fMyxyVFgXvmcVey1g@mail.gmail.com>

In order web telephone works correctly one should replace line 4

<script language="javascript" content-type="text/javascript"
src="http://ajax.googleapis.com/ajax/libs/jquery/1.4/jquery.min.js"></script>


for one referring local
jquery-1.4.2.js<http://fisheye.freeswitch.org/browse/freeswitch.git/clients/flex/jquery-1.4.2.js?r=25be760bbc8f18864d7961474665861dfb7d93ba>
which
is not used otherwise. Page is not working with line above.

I.e. it should be

<script language="javascript" content-type="text/javascript"
src="jquery-1.4.2.js
<http://ajax.googleapis.com/ajax/libs/jquery/1.4/jquery.min.js>"></script>


Regards,
   Dmitry.
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From sdame at 207me.com  Wed Jul 13 19:37:50 2011
From: sdame at 207me.com (Stephen Dame)
Date: Wed, 13 Jul 2011 11:37:50 -0400
Subject: [Freeswitch-dev] mod_rtmp
Message-ID: <000001cc4172$d36fd2e0$7a4f78a0$@com>



Great module. mod_rtmp   , 

 

I almost have it working.

 

I'm connecting to freeswitch thru mod_rtmp   using the flex client.  but
unclear to how the username password works for authentication.  Doesn't look
like anyplace in rtmp.conf.xml or the .html, or .mxml to set?

 

Using fs_cli -  It nets connects fine. looks like digits are received,  but
not sure how to configure or use default username or password so it can
login to make a call.

 

Also question on  rtmp://myserver/phone  should the profile in rtmp.conf.xml
match the rtmp name /phone?

 

Any tips appreciated.  Trying to get this working with BigBluebutton.

 

Will document a tutorial when finished.

 

Regards,

Stephen

 

 

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From yungwei at resolvity.com  Tue Jul 19 01:39:24 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 18 Jul 2011 17:39:24 -0400
Subject: [Freeswitch-dev] Regarding the patch for Multiple grammar ASR
	support
Message-ID: <33095823FD21DF429B481B5163264B7950CBC03BA1@VMBX102.ihostexchange.net>

Hi,

I am interested in taking advantages of the patch (http://jira.freeswitch.org/browse/FS-2906) for Multiple grammar ASR support from a javascript program.
But I am not sure exactly when to call the following function from a javascript program:
        ...
        this.addMultipleGrammars = function () {
                // hardcoded urls are just for my initial tests.
                this.session.execute("detect_speech", "grammar " + "http://192.168.16.9/pizza_arso.grxml");
                this.session.execute("detect_speech", "grammaron " + "http://192.168.16.9/pizza_arso.grxml");
        }
	  ...

I tried the following in scripts/js_modules/SpeechTools.jm, but this didn't work because the code snippet at the bottom prevented me from having multiple active grammars.
        .....
        while(this.asr.session.ready() && this.collected_index < this.req) {
                var x;
                this.needConfirm = false;
                if (!rv) {
                        this.asr.addMultipleGrammars();
                        rv = this.react(this.top_sound, this.top_sound);
                }
	  ...

I had to comment out the following piece of code in recog_asr_load_grammar function of /opt/freeswitch-snapshot/src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c to get my expected results.
This fix may not be needed. Perhaps you can point me to the right direction. Thanks.

        ...
        start_recognize = (char *) switch_core_hash_find(schannel->params, "start-recognize");
        if (zstr(start_recognize) || strcasecmp(start_recognize, "false"))
        {

		    // this disables all grammars
/*              switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "calling recog_channel_disable_all_grammars()\n");
                if (recog_channel_disable_all_grammars(schannel) != SWITCH_STATUS_SUCCESS) {
                        status = SWITCH_STATUS_FALSE;
                        goto done;
                }
*/
                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "calling recog_channel_enable_grammar()\n");
                if (recog_channel_enable_grammar(schannel, name) != SWITCH_STATUS_SUCCESS) {
                        status = SWITCH_STATUS_FALSE;
                        goto done;
                }
                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "calling recog_channel_start()\n");
                status = recog_channel_start(schannel);
        }



From msc at freeswitch.org  Mon Jul 25 21:48:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Jul 2011 10:48:30 -0700
Subject: [Freeswitch-dev] Bug in
	http://fisheye.freeswitch.org/browse/freeswitch.git/clients/flex/freeswitch.html
In-Reply-To: <CADDV4gwSKC8r7-tBCUYN6vwYCUzVET5S+fMyxyVFgXvmcVey1g@mail.gmail.com>
References: <CADDV4gwSKC8r7-tBCUYN6vwYCUzVET5S+fMyxyVFgXvmcVey1g@mail.gmail.com>
Message-ID: <CAKzWOxV3HKuROPYhRThA=ZiPNMepHc=xWqV0zQjBogUJMmZuWA@mail.gmail.com>

Can you open a case on jira.freeswitch.org so that we can keep track of this
issue properly?
Thanks,
MC

On Tue, Jul 19, 2011 at 4:40 AM, Dmitry Kravchenko <dimskraft at gmail.com>wrote:

> In order web telephone works correctly one should replace line 4
>
> <script language="javascript" content-type="text/javascript" src="http://ajax.googleapis.com/ajax/libs/jquery/1.4/jquery.min.js"></script>
>
>
> for one referring local jquery-1.4.2.js<http://fisheye.freeswitch.org/browse/freeswitch.git/clients/flex/jquery-1.4.2.js?r=25be760bbc8f18864d7961474665861dfb7d93ba> which
> is not used otherwise. Page is not working with line above.
>
> I.e. it should be
>
> <script language="javascript" content-type="text/javascript" src="jquery-1.4.2.js <http://ajax.googleapis.com/ajax/libs/jquery/1.4/jquery.min.js>"></script>
>
>
> Regards,
>    Dmitry.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Jul 27 20:05:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jul 2011 09:05:25 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxX1ObRXNcALC2gEk6BWhXxKCVeoDLXNrt+h1E3etN-F2w@mail.gmail.com>

Hey all,

We are having a simple FreeSWITCH conf call today:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_27

I want to talk about mod_fifo specifically and call queuing in general. If
you use mod_fifo at all please join us. There have been a lot of updates to
mod_fifo that make it more "ACD-like" but the wiki page needs some serious
attention. Please hop on and help us get the page updated.

Thanks!
-MC
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From marketing at cluecon.com  Wed Jul 27 20:36:50 2011
From: marketing at cluecon.com (Michael Collins)
Date: Wed, 27 Jul 2011 09:36:50 -0700
Subject: [Freeswitch-dev] ClueCon 2011 Update - Sofitel Full,
 Alternate Hotel Arrangements Being Made
Message-ID: <CAKzWOxWuMdEAAh9nn+mevEWwT74Ze-D8Wv9_GJjSr_GMpNSt2g@mail.gmail.com>

Hello!

Thank you all for supporting ClueCon 2011! We've maxed out the Sofitel, so
we're looking at another hotel for our attendees. If you are coming to
ClueCon and need a hotel room please contact Brian West ASAP at
marketing at cluecon.com or 877-742-CLUE. We need to know how many rooms we can
guarantee to another hotel in order to get discounted rates. Time is of the
essence, so please contact us right away.

Thanks!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From marketing at cluecon.com  Wed Jul 27 22:51:42 2011
From: marketing at cluecon.com (Michael Collins)
Date: Wed, 27 Jul 2011 11:51:42 -0700
Subject: [Freeswitch-dev] GOOD NEWS: Alternate ClueCon Hotel Arrangements
Message-ID: <CAKzWOxV4=az_Fcvx=6dLs_FuiLbYNd5VDaJD0DhEsvGp8ipXGg@mail.gmail.com>

Everyone say thanks to Brian West for making new hotel arrangements!  If you
need a room for ClueCon then we've got a great deal for you: $169 per night
and you still get the $699 ClueCon rate!

The new hotel:
The Talbott
20 E Delaware Place
Chicago, IL  60611
(312) 944-4970

Go get your room before we sell this one out, too! Thanks for supporting
ClueCon.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From marketing at cluecon.com  Wed Jul 27 23:58:42 2011
From: marketing at cluecon.com (Michael Collins)
Date: Wed, 27 Jul 2011 12:58:42 -0700
Subject: [Freeswitch-dev] ClueCon Hotel Update - Even *Better* Rates For The
	Talbott
Message-ID: <CAKzWOxViKtURs53AGNAoYKWPuUXqvHHB_+YaU8nYkYntNABGcw@mail.gmail.com>

Brian West is racking up the karma points today! He has finished talking to
The Talbott and has secured a rate of $148 per night! Everyone thank Brian
for his hard work on this - he's really outdone himself, especially when you
consider that Tony has been on vacation for more than a week.

If you're so inclined, you can visit Brian's wishlist
here<http://www.amazon.com/gp/registry/wishlist/1BWDJUX5LYQE0>.
:)

Thanks.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From marketing at cluecon.com  Thu Jul 28 01:55:16 2011
From: marketing at cluecon.com (Michael Collins)
Date: Wed, 27 Jul 2011 14:55:16 -0700
Subject: [Freeswitch-dev] FINAL UPDATE: ClueCon Hotel
Message-ID: <CAKzWOxWR+8HYOmU2qPQ=8ChWaEcUzd6xo2RAazOC9iJ8Sd=QPA@mail.gmail.com>

Thanks for your patience - we really wanted to make sure the word got out.
The updated information about the hotel can be found on our website:

http://cluecon.com/hotel

Keep in mind that you MUST ask for "in-house reservations" in order to get
to the right operator and get your special rate. Any questions please email
me off list.
Thanks!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From Matthew.Margolis at patlive.com  Tue Jul 26 22:36:53 2011
From: Matthew.Margolis at patlive.com (Matthew Margolis)
Date: Tue, 26 Jul 2011 14:36:53 -0400
Subject: [Freeswitch-dev] Wait times
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB604F30D6D@mail2.patlive.local>

Is there any way to set a fifo queue to automatically tell callers on
hold an approximate time they have left to wait every minute? Something
like "Thank you for holding, your current wait time is 'X' minutes."

 

If this feature doesn't exist, it would probably be very useful.

 

Matthew Margolis

Programmer
PATLive

T 850.766.3236

E matthew.margolis at patlive.com

 

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