From msc at freeswitch.org  Mon Mar  1 16:58:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 16:58:27 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call Agenda For Mar 3
Message-ID: <87f2f3b91003011658m724178d5i6409d75d3a1b5949@mail.gmail.com>

Hello folks!

I am trying to get a head start on the conference call agenda going out. I
have put a few things on the list already:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_03_03

I would like more people to add things. We had an impromptu mod_limit
discussion by Rupa S. last week and it was very cool. I recorded it and will
have it available for download shortly.

Last week's meeting was great! Please join us and bring your questions for
discussion about FreeSWITCH.

-Michael
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From dujinfang at gmail.com  Mon Mar  1 18:23:07 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 2 Mar 2010 10:23:07 +0800
Subject: [Freeswitch-dev] skypiax alerts on trunk code
Message-ID: <23f91031003011823v458a4ff7sbaa68fba46d3be97@mail.gmail.com>

Hi,

I'v seen lots improvements on trunk code, I just build a new server
with skype beta on FreeSWITCH Version 1.0.trunk (16848).  Ubuntu 8.04
64bit.

It works well. however I seen some messages on debug level console.

Locking assertion failure.  Backtrace:
#0 /usr/lib/libxcb-xlib.so.0 [0x7f1728a0c97c]
#1 /usr/lib/libxcb-xlib.so.0(xcb_xlib_unlock+0x24) [0x7f1728a0ca84]
#2 /usr/lib/libX11.so.6 [0x7f1728c54260]
#3 /usr/lib/libX11.so.6(XSync+0xd5) [0x7f1728c49915]
#4 /usr/local/freeswitch/mod/mod_skypiax.so [0x7f1728f1f2af]
#5 /usr/local/freeswitch/mod/mod_skypiax.so [0x7f1728f1fcf1]
#6 /usr/local/freeswitch/mod/mod_skypiax.so [0x7f1728f2353c]
#7 /usr/local/freeswitch/mod/mod_skypiax.so [0x7f1728f1bc7a]
#8 /lib/libpthread.so.0 [0x7f17346663ea]
#9 /lib/libc.so.6(clone+0x6d) [0x7f1734198cbd]
Locking assertion failure.  Backtrace:
#0 /usr/lib/libxcb-xlib.so.0 [0x7f1728a0c97c]
#1 /usr/lib/libxcb-xlib.so.0(xcb_xlib_lock+0x15) [0x7f1728a0ca15]
#2 /usr/lib/libX11.so.6 [0x7f1728c54323]
#3 /usr/lib/libX11.so.6 [0x7f1728c54a4b]
#4 /usr/lib/libX11.so.6 [0x7f1728c54d75]
#5 /usr/lib/libX11.so.6(XNextEvent+0x7f) [0x7f1728c3cc2f]
#6 /usr/local/freeswitch/mod/mod_skypiax.so [0x7f1728f1f79e]
#7 /lib/libpthread.so.0 [0x7f17346663ea]
#8 /lib/libc.so.6(clone+0x6d) [0x7f1734198cbd]


I will upgrade  and test later. and will gather more infos and will
report on jira if it's a bug or ignore if it's harmless.


From gmaruzz at celliax.org  Tue Mar  2 06:51:26 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 2 Mar 2010 15:51:26 +0100
Subject: [Freeswitch-dev] skypiax alerts on trunk code
In-Reply-To: <23f91031003011823v458a4ff7sbaa68fba46d3be97@mail.gmail.com>
References: <23f91031003011823v458a4ff7sbaa68fba46d3be97@mail.gmail.com>
Message-ID: <7b197bef1003020651w765b2a0au644fc0ec24d539a3@mail.gmail.com>

Hello Seven,

those are "almost unharmful" :).

Let me know how it goes your testing with skype beta.

In my experience skype beta client has problems in answering calls if
more channels share the same skypeusername and one channel is in a
call.

If you have any problem with mod_skypiax, before to delve in it, test
if the problem exist also with the stable, static build.

Thanks a lot for your testing

-giovanni

On Tue, Mar 2, 2010 at 3:23 AM, Seven Du <dujinfang at gmail.com> wrote:
> Hi,
>
> I'v seen lots improvements on trunk code, I just build a new server
> with skype beta on FreeSWITCH Version 1.0.trunk (16848). ?Ubuntu 8.04
> 64bit.
>
> It works well. however I seen some messages on debug level console.
>
> Locking assertion failure. ?Backtrace:
> #0 /usr/lib/libxcb-xlib.so.0 [0x7f1728a0c97c]
> #1 /usr/lib/libxcb-xlib.so.0(xcb_xlib_unlock+0x24) [0x7f1728a0ca84]
> #2 /usr/lib/libX11.so.6 [0x7f1728c54260]
> #3 /usr/lib/libX11.so.6(XSync+0xd5) [0x7f1728c49915]
> #4 /usr/local/freeswitch/mod/mod_skypiax.so [0x7f1728f1f2af]
> #5 /usr/local/freeswitch/mod/mod_skypiax.so [0x7f1728f1fcf1]
> #6 /usr/local/freeswitch/mod/mod_skypiax.so [0x7f1728f2353c]
> #7 /usr/local/freeswitch/mod/mod_skypiax.so [0x7f1728f1bc7a]
> #8 /lib/libpthread.so.0 [0x7f17346663ea]
> #9 /lib/libc.so.6(clone+0x6d) [0x7f1734198cbd]
> Locking assertion failure. ?Backtrace:
> #0 /usr/lib/libxcb-xlib.so.0 [0x7f1728a0c97c]
> #1 /usr/lib/libxcb-xlib.so.0(xcb_xlib_lock+0x15) [0x7f1728a0ca15]
> #2 /usr/lib/libX11.so.6 [0x7f1728c54323]
> #3 /usr/lib/libX11.so.6 [0x7f1728c54a4b]
> #4 /usr/lib/libX11.so.6 [0x7f1728c54d75]
> #5 /usr/lib/libX11.so.6(XNextEvent+0x7f) [0x7f1728c3cc2f]
> #6 /usr/local/freeswitch/mod/mod_skypiax.so [0x7f1728f1f79e]
> #7 /lib/libpthread.so.0 [0x7f17346663ea]
> #8 /lib/libc.so.6(clone+0x6d) [0x7f1734198cbd]
>
>
> I will upgrade ?and test later. and will gather more infos and will
> report on jira if it's a bug or ignore if it's harmless.
>
> _______________________________________________
> 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,

Giovanni Maruzzelli
Cell : +39-347-2665618


From m.sobkow at marketelsystems.com  Tue Mar  2 10:35:55 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 02 Mar 2010 12:35:55 -0600
Subject: [Freeswitch-dev] Bug with variable_endpoint_disposition and
	Answer-State event parameters
Message-ID: <4B8D5A8B.4010709@marketelsystems.com>

I think I may have found a bug in Freeswitch, as both answered and 
ringing calls have an "Answer-State" "answered" and 
variable_endpoint_disposition ANSWER, though in one case the call has 
NOT been answered.

For a normally answered call, the following Freeswitch events are 
propagated to Erlang:

CHANNEL_DATA
CHANNEL_PARK
CHANNEL_HANGUP
CHANNEL_UNPARK
CHANNEL_EXECUTE_COMPLETE
CHANNEL_STATE
CHANNEL_HANGUP_COMPLETE
CHANNEL_STATE
CHANNEL_DESTROY
CHANNEL_STATE

For an unanswered call, the events are:

CHANNEL_DATA
CHANNEL_EXECUTE
CHANNEL_PARK
CHANNEL_HANGUP
CHANNEL_UNPARK
CHANNEL_EXECUTE_COMPLETE
CHANNEL_STATE
CHANNEL_HANGUP_COMPLETE
CHANNEL_STATE
CHANNEL_DESTROY
CHANNEL_STATE

This is going to make the call processing a little complex, as both 
variants produce a CHANNEL_PARK early in the call processing, and later 
produce the HANGUP event.

The problem is that with an answered call, I have to uuid_bridge the 
operator on the CHANNEL_PARK event within 2 seconds to meet legal 
requirements, while with an unanswered call it's another 30 seconds 
(configurable) before the CHANNEL_HANGUP event propagates to let me know 
that the call went unanswered.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From m.sobkow at marketelsystems.com  Tue Mar  2 10:39:18 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 02 Mar 2010 12:39:18 -0600
Subject: [Freeswitch-dev] Bug with variable_endpoint_disposition and
 Answer-State event parameters
In-Reply-To: <4B8D5A8B.4010709@marketelsystems.com>
References: <4B8D5A8B.4010709@marketelsystems.com>
Message-ID: <4B8D5B56.9010708@marketelsystems.com>

Never mind.  What's actually happening is now that I've got the Erlang 
events firing, the timeout on the originate command doesn't seem to be 
working.  I had a timeout of 5 seconds set, but Freeswitch kept ringing 
until our voice mail system answered my phone (30 seconds.)

*sigh*

Not that it's any better a result, as we DO need to timeout calls that 
aren't being answered.

Mark Sobkow wrote:
> I think I may have found a bug in Freeswitch, as both answered and 
> ringing calls have an "Answer-State" "answered" and 
> variable_endpoint_disposition ANSWER, though in one case the call has 
> NOT been answered.
>
> For a normally answered call, the following Freeswitch events are 
> propagated to Erlang:
>
> CHANNEL_DATA
> CHANNEL_PARK
> CHANNEL_HANGUP
> CHANNEL_UNPARK
> CHANNEL_EXECUTE_COMPLETE
> CHANNEL_STATE
> CHANNEL_HANGUP_COMPLETE
> CHANNEL_STATE
> CHANNEL_DESTROY
> CHANNEL_STATE
>
> For an unanswered call, the events are:
>
> CHANNEL_DATA
> CHANNEL_EXECUTE
> CHANNEL_PARK
> CHANNEL_HANGUP
> CHANNEL_UNPARK
> CHANNEL_EXECUTE_COMPLETE
> CHANNEL_STATE
> CHANNEL_HANGUP_COMPLETE
> CHANNEL_STATE
> CHANNEL_DESTROY
> CHANNEL_STATE
>
> This is going to make the call processing a little complex, as both 
> variants produce a CHANNEL_PARK early in the call processing, and 
> later produce the HANGUP event.
>
> The problem is that with an answered call, I have to uuid_bridge the 
> operator on the CHANNEL_PARK event within 2 seconds to meet legal 
> requirements, while with an unanswered call it's another 30 seconds 
> (configurable) before the CHANNEL_HANGUP event propagates to let me 
> know that the call went unanswered.
>


-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From andrew at hijacked.us  Tue Mar  2 10:45:13 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 2 Mar 2010 13:45:13 -0500
Subject: [Freeswitch-dev] Bug with variable_endpoint_disposition and
	Answer-State event parameters
In-Reply-To: <4B8D5B56.9010708@marketelsystems.com>
References: <4B8D5A8B.4010709@marketelsystems.com>
	<4B8D5B56.9010708@marketelsystems.com>
Message-ID: <20100302184512.GF1751@hijacked.us>

On Tue, Mar 02, 2010 at 12:39:18PM -0600, Mark Sobkow wrote:
> Never mind.  What's actually happening is now that I've got the Erlang 
> events firing, the timeout on the originate command doesn't seem to be 
> working.  I had a timeout of 5 seconds set, but Freeswitch kept ringing 
> until our voice mail system answered my phone (30 seconds.)
> 
> *sigh*
> 
> Not that it's any better a result, as we DO need to timeout calls that 
> aren't being answered.
>
Which timeout variable(s) are you setting, there are several and they
have different effects. See
http://wiki.freeswitch.org/wiki/Channel_Variables for at least a partial
list.

Andrew


From anthony.minessale at gmail.com  Tue Mar  2 10:47:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 12:47:02 -0600
Subject: [Freeswitch-dev] Bug with variable_endpoint_disposition and
	Answer-State event parameters
In-Reply-To: <4B8D5B56.9010708@marketelsystems.com>
References: <4B8D5A8B.4010709@marketelsystems.com>
	<4B8D5B56.9010708@marketelsystems.com>
Message-ID: <191c3a031003021047y26294074rb164e11e39c76c6a@mail.gmail.com>

Clearly you used the originate timeout incorrectly.


On Tue, Mar 2, 2010 at 12:39 PM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

> Never mind.  What's actually happening is now that I've got the Erlang
> events firing, the timeout on the originate command doesn't seem to be
> working.  I had a timeout of 5 seconds set, but Freeswitch kept ringing
> until our voice mail system answered my phone (30 seconds.)
>
> *sigh*
>
> Not that it's any better a result, as we DO need to timeout calls that
> aren't being answered.
>
> Mark Sobkow wrote:
> > I think I may have found a bug in Freeswitch, as both answered and
> > ringing calls have an "Answer-State" "answered" and
> > variable_endpoint_disposition ANSWER, though in one case the call has
> > NOT been answered.
> >
> > For a normally answered call, the following Freeswitch events are
> > propagated to Erlang:
> >
> > CHANNEL_DATA
> > CHANNEL_PARK
> > CHANNEL_HANGUP
> > CHANNEL_UNPARK
> > CHANNEL_EXECUTE_COMPLETE
> > CHANNEL_STATE
> > CHANNEL_HANGUP_COMPLETE
> > CHANNEL_STATE
> > CHANNEL_DESTROY
> > CHANNEL_STATE
> >
> > For an unanswered call, the events are:
> >
> > CHANNEL_DATA
> > CHANNEL_EXECUTE
> > CHANNEL_PARK
> > CHANNEL_HANGUP
> > CHANNEL_UNPARK
> > CHANNEL_EXECUTE_COMPLETE
> > CHANNEL_STATE
> > CHANNEL_HANGUP_COMPLETE
> > CHANNEL_STATE
> > CHANNEL_DESTROY
> > CHANNEL_STATE
> >
> > This is going to make the call processing a little complex, as both
> > variants produce a CHANNEL_PARK early in the call processing, and
> > later produce the HANGUP event.
> >
> > The problem is that with an answered call, I have to uuid_bridge the
> > operator on the CHANNEL_PARK event within 2 seconds to meet legal
> > requirements, while with an unanswered call it's another 30 seconds
> > (configurable) before the CHANNEL_HANGUP event propagates to let me
> > know that the call went unanswered.
> >
>
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From m.sobkow at marketelsystems.com  Tue Mar  2 10:47:44 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 02 Mar 2010 12:47:44 -0600
Subject: [Freeswitch-dev] Bug with variable_endpoint_disposition and
 Answer-State event parameters
In-Reply-To: <4B8D5B56.9010708@marketelsystems.com>
References: <4B8D5A8B.4010709@marketelsystems.com>
	<4B8D5B56.9010708@marketelsystems.com>
Message-ID: <4B8D5D50.8020309@marketelsystems.com>

I tried changing the answer timeout to 2 seconds, but Freeswitch 
"originate" is definitely not honouring that request.  The originate 
command is:

originate, sofia/external/533 at rats.marketel '&erlang( pbx_called_cust 
pursuit at testsrv )' 5

Mark Sobkow wrote:
> Never mind.  What's actually happening is now that I've got the Erlang 
> events firing, the timeout on the originate command doesn't seem to be 
> working.  I had a timeout of 5 seconds set, but Freeswitch kept 
> ringing until our voice mail system answered my phone (30 seconds.)
>
> *sigh*
>
> Not that it's any better a result, as we DO need to timeout calls that 
> aren't being answered.
>
> Mark Sobkow wrote:
>> I think I may have found a bug in Freeswitch, as both answered and 
>> ringing calls have an "Answer-State" "answered" and 
>> variable_endpoint_disposition ANSWER, though in one case the call has 
>> NOT been answered.
>>
>> For a normally answered call, the following Freeswitch events are 
>> propagated to Erlang:
>>
>> CHANNEL_DATA
>> CHANNEL_PARK
>> CHANNEL_HANGUP
>> CHANNEL_UNPARK
>> CHANNEL_EXECUTE_COMPLETE
>> CHANNEL_STATE
>> CHANNEL_HANGUP_COMPLETE
>> CHANNEL_STATE
>> CHANNEL_DESTROY
>> CHANNEL_STATE
>>
>> For an unanswered call, the events are:
>>
>> CHANNEL_DATA
>> CHANNEL_EXECUTE
>> CHANNEL_PARK
>> CHANNEL_HANGUP
>> CHANNEL_UNPARK
>> CHANNEL_EXECUTE_COMPLETE
>> CHANNEL_STATE
>> CHANNEL_HANGUP_COMPLETE
>> CHANNEL_STATE
>> CHANNEL_DESTROY
>> CHANNEL_STATE
>>
>> This is going to make the call processing a little complex, as both 
>> variants produce a CHANNEL_PARK early in the call processing, and 
>> later produce the HANGUP event.
>>
>> The problem is that with an answered call, I have to uuid_bridge the 
>> operator on the CHANNEL_PARK event within 2 seconds to meet legal 
>> requirements, while with an unanswered call it's another 30 seconds 
>> (configurable) before the CHANNEL_HANGUP event propagates to let me 
>> know that the call went unanswered.
>>
>
>


-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From anthony.minessale at gmail.com  Tue Mar  2 10:57:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 12:57:06 -0600
Subject: [Freeswitch-dev] Bug with variable_endpoint_disposition and
	Answer-State event parameters
In-Reply-To: <4B8D5D50.8020309@marketelsystems.com>
References: <4B8D5A8B.4010709@marketelsystems.com>
	<4B8D5B56.9010708@marketelsystems.com>
	<4B8D5D50.8020309@marketelsystems.com>
Message-ID: <191c3a031003021057o3a4a2815ncf4f1b58c37f2b77@mail.gmail.com>

do you mean bridge_answer_timeout?

if you see the wiki, that variable only pertains to bridged calls in
early_media state.

you probably want:

http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout



On Tue, Mar 2, 2010 at 12:47 PM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

> I tried changing the answer timeout to 2 seconds, but Freeswitch
> "originate" is definitely not honouring that request.  The originate
> command is:
>
> originate, sofia/external/533 at rats.marketel '&erlang( pbx_called_cust
> pursuit at testsrv )' 5
>
> Mark Sobkow wrote:
> > Never mind.  What's actually happening is now that I've got the Erlang
> > events firing, the timeout on the originate command doesn't seem to be
> > working.  I had a timeout of 5 seconds set, but Freeswitch kept
> > ringing until our voice mail system answered my phone (30 seconds.)
> >
> > *sigh*
> >
> > Not that it's any better a result, as we DO need to timeout calls that
> > aren't being answered.
> >
> > Mark Sobkow wrote:
> >> I think I may have found a bug in Freeswitch, as both answered and
> >> ringing calls have an "Answer-State" "answered" and
> >> variable_endpoint_disposition ANSWER, though in one case the call has
> >> NOT been answered.
> >>
> >> For a normally answered call, the following Freeswitch events are
> >> propagated to Erlang:
> >>
> >> CHANNEL_DATA
> >> CHANNEL_PARK
> >> CHANNEL_HANGUP
> >> CHANNEL_UNPARK
> >> CHANNEL_EXECUTE_COMPLETE
> >> CHANNEL_STATE
> >> CHANNEL_HANGUP_COMPLETE
> >> CHANNEL_STATE
> >> CHANNEL_DESTROY
> >> CHANNEL_STATE
> >>
> >> For an unanswered call, the events are:
> >>
> >> CHANNEL_DATA
> >> CHANNEL_EXECUTE
> >> CHANNEL_PARK
> >> CHANNEL_HANGUP
> >> CHANNEL_UNPARK
> >> CHANNEL_EXECUTE_COMPLETE
> >> CHANNEL_STATE
> >> CHANNEL_HANGUP_COMPLETE
> >> CHANNEL_STATE
> >> CHANNEL_DESTROY
> >> CHANNEL_STATE
> >>
> >> This is going to make the call processing a little complex, as both
> >> variants produce a CHANNEL_PARK early in the call processing, and
> >> later produce the HANGUP event.
> >>
> >> The problem is that with an answered call, I have to uuid_bridge the
> >> operator on the CHANNEL_PARK event within 2 seconds to meet legal
> >> requirements, while with an unanswered call it's another 30 seconds
> >> (configurable) before the CHANNEL_HANGUP event propagates to let me
> >> know that the call went unanswered.
> >>
> >
> >
>
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From m.sobkow at marketelsystems.com  Tue Mar  2 11:07:56 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 02 Mar 2010 13:07:56 -0600
Subject: [Freeswitch-dev] Bug with variable_endpoint_disposition and
 Answer-State event parameters
In-Reply-To: <20100302184512.GF1751@hijacked.us>
References: <4B8D5A8B.4010709@marketelsystems.com>
	<4B8D5B56.9010708@marketelsystems.com>
	<20100302184512.GF1751@hijacked.us>
Message-ID: <4B8D620C.9030902@marketelsystems.com>

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From anthony.minessale at gmail.com  Tue Mar  2 15:07:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 17:07:42 -0600
Subject: [Freeswitch-dev] When is 1.0.5 due out?
In-Reply-To: <4ABCE5DD.30903@navynet.it>
References: <4ABBCF9E.6040408@marketelsystems.com> <4ABCE5DD.30903@navynet.it>
Message-ID: <191c3a031003021507x7b194065p20a5a537e9555e62@mail.gmail.com>

Hey,

Where have you been hiding?

We are going to add t38 to FS with coppice and I need something to test
against.
Does that test web page that sends a fax still work?



On Fri, Sep 25, 2009 at 9:46 AM, Massimo "CtRiX" Cetra
<ctrix+fs at navynet.it<ctrix%2Bfs at navynet.it>
> wrote:

> Mark Sobkow ha scritto:
> > Alternatively, if someone could point me to a repo with control files
> > for 1.0.3 or 1.0.4, we could do our own source builds and packaging.
> >
> Freeswitch code actually contains a debian/ dir with everything needed
> to package it the debian way.
>
> Max
>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From peder at networkoblivion.com  Tue Mar  2 15:43:50 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 2 Mar 2010 17:43:50 -0600
Subject: [Freeswitch-dev] When is 1.0.5 due out?
In-Reply-To: <191c3a031003021507x7b194065p20a5a537e9555e62@mail.gmail.com>
References: <4ABBCF9E.6040408@marketelsystems.com> <4ABCE5DD.30903@navynet.it>
	<191c3a031003021507x7b194065p20a5a537e9555e62@mail.gmail.com>
Message-ID: <0afc01caba62$35adbba0$a10932e0$@com>

You can use this to send faxes.  It isn't immediate, but it is free:

 

http://faxzero.com/

 

 

 

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Tuesday, March 02, 2010 5:08 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] When is 1.0.5 due out?

 

Hey,

Where have you been hiding?

We are going to add t38 to FS with coppice and I need something to test
against.
Does that test web page that sends a fax still work?




On Fri, Sep 25, 2009 at 9:46 AM, Massimo "CtRiX" Cetra <ctrix+fs at navynet.it
<mailto:ctrix%2Bfs at navynet.it> > wrote:

Mark Sobkow ha scritto:

> Alternatively, if someone could point me to a repo with control files
> for 1.0.3 or 1.0.4, we could do our own source builds and packaging.
>

Freeswitch code actually contains a debian/ dir with everything needed
to package it the debian way.

Max



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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From msc at freeswitch.org  Wed Mar  3 08:19:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Mar 2010 08:19:11 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Agenda
Message-ID: <87f2f3b91003030819ld02171ub6bc4cbc191eeb94@mail.gmail.com>

Hello all. We have a nice agenda today:

http://bit.ly/9zDiBq

We invite everyone to join us. I believe that today we may have user todpunk
giving a talk on mod_fifo.

Talk to you all soon,
Michael
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From Suneel.Papineni at mettoni.com  Fri Mar  5 03:04:05 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 5 Mar 2010 11:04:05 -0000
Subject: [Freeswitch-dev] Call transfer using eventsockets with FSComm
Message-ID: <3181A30B8C35AB4AA8577B78DDF46138068F6214@nickel.mettonigroup.com>

Hi,

 

Is there a way to transfer call with FSComm. I tried different ways
using eventsockets but failed.

 

Tried the scenario as follows:

Received a call from 1001 to FSComm (registered with 1002) and is
answered (Unique-id is 784dd690-6d0a-47de-b5f4-b923264581a5). Made a
call from FSComm to extension 1003 and is answered (Unique-id is
ec7a3bdd-c265-41fb-8cef-b897d2e8bf62). Now I want to transfer this call
to 1001 and FSComm should be out of loop. 

 

a.       Tried with command "api uuid_transfer
784dd690-6d0a-47de-b5f4-b923264581a5 -both park inline" but failed. With
this call at 1001 is dropped and call between FSComm and 1003 is still
there.

b.      Tried with command "api uuid_bridge
784dd690-6d0a-47de-b5f4-b923264581a5
ec7a3bdd-c265-41fb-8cef-b897d2e8bf62". With this all calls are dropped.

 

Could someone let me know if there is any procedure for call transfer
(using event sockets).

 

Thanks & Regards

Suneel


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
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From jmesquita at freeswitch.org  Sun Mar  7 20:14:36 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 8 Mar 2010 01:14:36 -0300
Subject: [Freeswitch-dev] Call transfer using eventsockets with FSComm
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138068F6214@nickel.mettonigroup.com>
References: <Acq8U5J/iQUC0nAxSbWQSsxFzefS/Q==>
	<3181A30B8C35AB4AA8577B78DDF46138068F6214@nickel.mettonigroup.com>
Message-ID: <ebbc2d551003072014h4368cb2x1a02c0fb5953504d@mail.gmail.com>

A few logs for this would be just amazing so I can help figure this out. I
am very interested at this process because when I can give FSComm some love,
I will start implementing those types of features.

Regards,
Jo?o Mesquita


On Fri, Mar 5, 2010 at 8:04 AM, Suneel Papineni <Suneel.Papineni at mettoni.com
> wrote:

>  Hi,
>
>
>
> Is there a way to transfer call with FSComm. I tried different ways using
> eventsockets but failed.
>
>
>
> Tried the scenario as follows:
>
> Received a call from 1001 to FSComm (registered with 1002) and is answered
> (Unique-id is 784dd690-6d0a-47de-b5f4-b923264581a5). Made a call from FSComm
> to extension 1003 and is answered (Unique-id is
> ec7a3bdd-c265-41fb-8cef-b897d2e8bf62). Now I want to transfer this call to
> 1001 and FSComm should be out of loop.
>
>
>
> a.       Tried with command ?api uuid_transfer
> 784dd690-6d0a-47de-b5f4-b923264581a5 ?both park inline? but failed. With
> this call at 1001 is dropped and call between FSComm and 1003 is still
> there.
>
> b.      Tried with command ?api uuid_bridge
> 784dd690-6d0a-47de-b5f4-b923264581a5 ec7a3bdd-c265-41fb-8cef-b897d2e8bf62?.
> With this all calls are dropped.
>
>
>
> Could someone let me know if there is any procedure for call transfer
> (using event sockets).
>
>
>
> Thanks & Regards
>
> Suneel
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
>
> _______________________________________________
> 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 jmesquita at freeswitch.org  Sun Mar  7 20:18:59 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 8 Mar 2010 01:18:59 -0300
Subject: [Freeswitch-dev] FSComm basic issue
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613806886903@nickel.mettonigroup.com>
References: <Acq3BRPNPSeAcwujTl+n9yQujbXGcA==>
	<3181A30B8C35AB4AA8577B78DDF4613806886903@nickel.mettonigroup.com>
Message-ID: <ebbc2d551003072018r294ee37ag4774121fea30e689@mail.gmail.com>

Sorry for the late reply. Was out of town...

Inline...

JM

On Fri, Feb 26, 2010 at 1:59 PM, Suneel Papineni <
Suneel.Papineni at mettoni.com> wrote:

>  Hi,
>
>
>
> I am trying to use FSComm with Freeswitch and facing following issues.
>
>
>
> 1.       Using pre-build binary (windows), when the application is started
> FSComm is getting Registered properly. When I tried to make a call, UI
> displays Dialing... but unable to see any SIP (INVITE) messages in wireshark
> traces. After sometime UI displays with message ?Call with (destination
> number) failed with reason DESTINATION_OUT_OF_ORDER though destination
> number is registered with another FSComm?
>

How exactly are you dialing from the UI? What version are you running? You
can check the build on Help -> About

> 2.       Also I am unable to see any logs generated in the log folder.
>
>
>
What do you mean by unable?

> Downloaded the latest source code (Freeswitch 1.0.5 latest updated as on
> 26/02/10 at 4am) and tried to build FSComm. Build was succeeded. Application
> (FSComm) also started and displayed with UI. When I try to change the
> preferences, it has thrown Porta Audio Error saying ?Error Querying Audio
> Devices? even though proper audio devices are present.
>

Have you checked if mod_portaudio was loaded? I still have to implement some
error messages on startup when PA or the core does not load.


> Also it doesn?t create folders like ?conf?, ?mod?. Even after copying all
> the required dll?s and mod files (as specified in FSComm wiki pages),
> application is throwing the same error.
>
That's why ppl have been packaging the Windows build. On Linux, we create
all dirs just fine. I just checked.


>
>
> I am using Windows XP machine. Built a Debug & Release version with 32-bit
> option.
>
> If someone has built FSComm for windows environment and is working fine,
> could you please let me know if there are any additional things I need to do
> to make it work.
>

I will try to get myself built FSComm over Windows, but I have lousy skills
on MSVS... jlenk was helping me out, will pick it up again to get code
improved.

>
>
> Thanks & Regards
>
> Suneel
>
>
>
>
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
>
> _______________________________________________
> 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  Mon Mar  8 05:30:31 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Mon, 8 Mar 2010 14:30:31 +0100
Subject: [Freeswitch-dev] mod lua session:bridged() not implemented?
Message-ID: <50e456911003080530n4d02bee3y1d444e89719eb35a@mail.gmail.com>

Hi Guys,
Freeswitch rocks!

According to the wiki page I could use (session:bridged() to check if
session is bridged...
http://wiki.freeswitch.org/wiki/Lua#session:bridged
if (session:bridged() == true) do
    -- Do something
end
But in latest trunk session:bridged() retuns nil...

If I list the session functions like this:
http://wiki.freeswitch.org/wiki/Lua#How_can_I_find_useful_undocumented_Session_Functions.3F
session:bridged is not in the list...
Do you plan to implement this function?

How can I detect a bridge on the other leg in lua?

I need this to create an automatic predictive dialer, which dials
customers and if the customer picks up the phone the lua script plays
a message and transfers the other leg to an operator...
I start the call using event-socket api originate and bridges the
customer with a loopback lua script:
"api originate {originate_timeout=30,origination_caller_id_number=3812345678}sofia/gateway/provider/12345789
'&lua(cc.lua predictive)'"
In the lua script I would have to wait for the a-leg to bridge, how to
achive this without session:bridged()?


From gmaruzz at celliax.org  Mon Mar  8 06:46:45 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 8 Mar 2010 15:46:45 +0100
Subject: [Freeswitch-dev] mod_skypiax (skype endpoint) call for tests
Message-ID: <7b197bef1003080646q91b6e25td5c719739ad8b5b4@mail.gmail.com>

Hello FreeSWITCHers,

it would be very useful if you could update to the latest svn and test
mod_skypiax in your use cases, and report any problems.

Please note that the wiki page http://wiki.freeswitch.org/wiki/Skypiax
has been almost completely revamped, with complete coverage of the new
options, channel variables, command line commands, events,
installation, etc. Take the time to at least browse through it,
because much new material has been added, that can spare you time and
problems.

Would be very useful if you report anything wrong, missing, or hard to
understand in the wiki page itself too (btw, English is not my first
language ;) ).

Thanks in advance for your cooperation, that makes this community so
nice a place to be in.

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From msc at freeswitch.org  Mon Mar  8 09:33:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 09:33:42 -0800
Subject: [Freeswitch-dev] mod lua session:bridged() not implemented?
In-Reply-To: <50e456911003080530n4d02bee3y1d444e89719eb35a@mail.gmail.com>
References: <50e456911003080530n4d02bee3y1d444e89719eb35a@mail.gmail.com>
Message-ID: <87f2f3b91003080933x754fb560v547a88d931467070@mail.gmail.com>

I think you might be looking at the wrong tool. The session:bridged() method
is for knowing if the current session is bridged to another leg. Tony says
it has a very specific use case and I doubt that this is what they had in
mind.

If you're building a predictive dialer and you are generating the outbound
call leg then you're most likely in need of knowing that the far end has
answered. In a case like this I would build a dialplan extension and have
the b leg drop in there. Then set execute_on_answer to your lua script. You
could also use that dialplan to handle timeouts, call failures, etc. This
way you're letting the dialplan do what it's good at and your Lua script can
focus on the single task of handling an answered call...

-MC

On Mon, Mar 8, 2010 at 5:30 AM, rentmycoder rentmycoder <
rentmycoder at gmail.com> wrote:

> Hi Guys,
> Freeswitch rocks!
>
> According to the wiki page I could use (session:bridged() to check if
> session is bridged...
> http://wiki.freeswitch.org/wiki/Lua#session:bridged
> if (session:bridged() == true) do
>    -- Do something
> end
> But in latest trunk session:bridged() retuns nil...
>
> If I list the session functions like this:
>
> http://wiki.freeswitch.org/wiki/Lua#How_can_I_find_useful_undocumented_Session_Functions.3F
> session:bridged is not in the list...
> Do you plan to implement this function?
>
> How can I detect a bridge on the other leg in lua?
>
> I need this to create an automatic predictive dialer, which dials
> customers and if the customer picks up the phone the lua script plays
> a message and transfers the other leg to an operator...
> I start the call using event-socket api originate and bridges the
> customer with a loopback lua script:
> "api originate
> {originate_timeout=30,origination_caller_id_number=3812345678}sofia/gateway/provider/12345789
> '&lua(cc.lua predictive)'"
> In the lua script I would have to wait for the a-leg to bridge, how to
> achive this without session:bridged()?
>
> _______________________________________________
> 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  Mon Mar  8 10:06:17 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Mon, 8 Mar 2010 19:06:17 +0100
Subject: [Freeswitch-dev] mod lua session:bridged() not implemented?
Message-ID: <50e456911003081006g7ce1151fo285660906cdbaab7@mail.gmail.com>

Yes you are absolutelly right, maybe I might be looking at the wrong tool...
I have tried several approaches but haven't found the right one...
I don't only need to detect if a call was answered, but I need also to
detect and log the failure cause
in case the call was not answered. Which usually can be found in the
originate_disposition variable...

If I use a socket command:
"bgapi originate
{ignore_early_media=true,originate_timeout=10,api_hangup_hook='luarun
cc.lua predictive_hangup'}sofia/gateway/gwname/13246789 9999"
I cannot detect the originate_disposition variable in
predictive_hangup() in lua, becouse the session is nil after hangup...

I think I should start dialing out from the dialplan, but in this case
how to call to the dialplan using sockets?
I've tried:
"bgapi originate {phoneno=13246789}loopback/9999"
with no success...

What do you suggest?
Is it possible to get this working only with 2 call legs???




I think you might be looking at the wrong tool. The session:bridged() method
is for knowing if the current session is bridged to another leg. Tony says
it has a very specific use case and I doubt that this is what they had in
mind.

If you're building a predictive dialer and you are generating the outbound
call leg then you're most likely in need of knowing that the far end has
answered. In a case like this I would build a dialplan extension and have
the b leg drop in there. Then set execute_on_answer to your lua script. You
could also use that dialplan to handle timeouts, call failures, etc. This
way you're letting the dialplan do what it's good at and your Lua script can
focus on the single task of handling an answered call...

-MC


From Suneel.Papineni at mettoni.com  Tue Mar  9 02:44:10 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Tue, 9 Mar 2010 10:44:10 -0000
Subject: [Freeswitch-dev] FSComm basic issue
In-Reply-To: <ebbc2d551003072018r294ee37ag4774121fea30e689@mail.gmail.com>
References: <Acq3BRPNPSeAcwujTl+n9yQujbXGcA==><3181A30B8C35AB4AA8577B78DDF4613806886903@nickel.mettonigroup.com>
	<ebbc2d551003072018r294ee37ag4774121fea30e689@mail.gmail.com>
Message-ID: <3181A30B8C35AB4AA8577B78DDF46138068F6893@nickel.mettonigroup.com>

Hi Jo,

 

This time I got the logs for call transfer issue... FYI, I am running the application in Windows XP system. I have connected to FSComm through Event Socket connection and sending commands as there is no option with FSComm as of now.

 

a.       For the first scenario, I made a call from 33004 (an extension) to FSComm (registered as 33001) and answered. Now from FSComm another call is made to an extension 35011. With this 33004 was put on music on hold and call is there between FSComm and 35011. I want to transfer this call, so that call will be there between 33004 and 35011. (FSComm will be out of loop) For this I issued the following command "api uuid_transfer b50b2200-9928-4a4c-8198-4b61b69fb38f - both park inline" After issuing this command call transfer is not done. Active call was in between FSComm and 35011. Music on Hold on 33004 was stopped, but call was not disconnected (When I checked with "pa list" command, it shown call with 33004 is on hold). Please find attached log for this in Freeswitch_uuid_transfer.log

b.      In the second scenario, I issues bridge command as "api uuid_bridge <Unique-id of first call between 33004 and FSComm> <Unique-id of second call between FSComm and 35011>. After issuing this command, Call between FSComm and 35011 is disconnected and active call is there between FSComm and 33004. Respective logs are attached (Freeswitch_uuid_bridge.log)

 

Please let me know if I need to change some thing at dial-plan in Freeswitch.xml

 

Right now I added the "red" colored lines in the dial-plan (I think I am wrong here). Please let me know if this is wrong and suggest me what needs to done. 

 

                        <extension name="public_extensions">

                                <condition field="destination_number" expression="^(.*)$">

                                        <action application="export" data="origination_caller_id_number=${fscomm_caller_id_num}"/>

                                        <action application="export" data="origination_caller_id_name=${fscomm_caller_id_name}"/>

                                        <action application="info"/>

                                        <action application="set" data="ringback=${us-ring}"/>

                                        <action application="pre_answer"/>

                                        <action application="set" data="park_after_bridge=true"/>

                                        <action application="bridge" data="portaudio"/>

                                        <action application="set" data="transfer_after_bridge=1000"/>

                                        <action application="set" data="hangup_after_bridge=true"/>

                                </condition>

                        </extension>

 

 

Thanks & Regards

Suneel

 

 

From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita
Sent: 08 March 2010 04:19
To: freeswitch-dev at lists.freeswitch.org
Cc: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] FSComm basic issue

 

Sorry for the late reply. Was out of town...

 

Inline...

JM

On Fri, Feb 26, 2010 at 1:59 PM, Suneel Papineni <Suneel.Papineni at mettoni.com> wrote:

Hi,

 

I am trying to use FSComm with Freeswitch and facing following issues. 

 

1.       Using pre-build binary (windows), when the application is started FSComm is getting Registered properly. When I tried to make a call, UI displays Dialing... but unable to see any SIP (INVITE) messages in wireshark traces. After sometime UI displays with message "Call with (destination number) failed with reason DESTINATION_OUT_OF_ORDER though destination number is registered with another FSComm"

 

How exactly are you dialing from the UI? What version are you running? You can check the build on Help -> About 

	2.       Also I am unable to see any logs generated in the log folder.

	 

What do you mean by unable? 

	Downloaded the latest source code (Freeswitch 1.0.5 latest updated as on 26/02/10 at 4am) and tried to build FSComm. Build was succeeded. Application (FSComm) also started and displayed with UI. When I try to change the preferences, it has thrown Porta Audio Error saying "Error Querying Audio Devices" even though proper audio devices are present.

 

Have you checked if mod_portaudio was loaded? I still have to implement some error messages on startup when PA or the core does not load.

 

	Also it doesn't create folders like "conf", "mod". Even after copying all the required dll's and mod files (as specified in FSComm wiki pages), application is throwing the same error.

That's why ppl have been packaging the Windows build. On Linux, we create all dirs just fine. I just checked.

 

	 

	I am using Windows XP machine. Built a Debug & Release version with 32-bit option.

	If someone has built FSComm for windows environment and is working fine, could you please let me know if there are any additional things I need to do to make it work.

 

I will try to get myself built FSComm over Windows, but I have lousy skills on MSVS... jlenk was helping me out, will pick it up again to get code improved.

	 

	Thanks & Regards

	Suneel

	 

	 

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	intended solely for the use of the individual or entity to whom they
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	the system manager.  http://www.mettoni.com
	 
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	Registered in England and Wales: 4485956
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	*************************************************************************

	
	_______________________________________________
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Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
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From brian at freeswitch.org  Tue Mar  9 06:21:58 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Mar 2010 08:21:58 -0600
Subject: [Freeswitch-dev] FSComm basic issue
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138068F6893@nickel.mettonigroup.com>
References: <Acq3BRPNPSeAcwujTl+n9yQujbXGcA==><3181A30B8C35AB4AA8577B78DDF4613806886903@nickel.mettonigroup.com>
	<ebbc2d551003072018r294ee37ag4774121fea30e689@mail.gmail.com>
	<3181A30B8C35AB4AA8577B78DDF46138068F6893@nickel.mettonigroup.com>
Message-ID: <A86CCA29-0C39-4F06-8B59-4FDCB5476F0E@freeswitch.org>

All this goes on jira.freeswitch.org under FSCOMM.

/b

On Mar 9, 2010, at 4:44 AM, Suneel Papineni wrote:

> Hi Jo,

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From m.sobkow at marketelsystems.com  Tue Mar  9 08:12:47 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Mar 2010 10:12:47 -0600
Subject: [Freeswitch-dev] Problem with call recording
Message-ID: <4B96737F.9030305@marketelsystems.com>

I'm having a problem with call recording.  I tell Freeswitch to record a 
call to a file, and it responds ok, and the file shows up on the file 
system.  I later tell Freeswitch to stop recording, and it returns ok, 
but the file keeps growing as it doesn't actually stop recording.

As a result, any subsequent attempts to start recording the session to a 
different file name fails.

In my case, an operator is logged in to the phone system and receiving 
calls from customers or from automated callbacks (if the customer didn't 
want to wait.)  So I don't need to record the entire operator session, 
just _segments_ of it.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From brian at freeswitch.org  Tue Mar  9 08:17:42 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Mar 2010 10:17:42 -0600
Subject: [Freeswitch-dev] Problem with call recording
In-Reply-To: <4B96737F.9030305@marketelsystems.com>
References: <4B96737F.9030305@marketelsystems.com>
Message-ID: <8D801943-7D35-4A54-8B26-791AE3333605@freeswitch.org>

Mark,
	Bug reports belong on Jira.

	http://jira.freeswitch.org


/b

On Mar 9, 2010, at 10:12 AM, Mark Sobkow wrote:

> I'm having a problem with call recording.  I tell Freeswitch to record a 
> call to a file, and it responds ok, and the file shows up on the file 
> system.  I later tell Freeswitch to stop recording, and it returns ok, 
> but the file keeps growing as it doesn't actually stop recording.
> 
> As a result, any subsequent attempts to start recording the session to a 
> different file name fails.
> 
> In my case, an operator is logged in to the phone system and receiving 
> calls from customers or from automated callbacks (if the customer didn't 
> want to wait.)  So I don't need to record the entire operator session, 
> just _segments_ of it.



From brian at freeswitch.org  Tue Mar  9 08:48:47 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Mar 2010 10:48:47 -0600
Subject: [Freeswitch-dev] Problem with call recording
In-Reply-To: <4B96737F.9030305@marketelsystems.com>
References: <4B96737F.9030305@marketelsystems.com>
Message-ID: <029D61D2-0C59-4087-BFD6-814DC1988A56@freeswitch.org>

Please provide the exact steps you're doing to reproduce this.  I'm doing this with uuid_record and starting and stopping it fine and it doesn't error out.  So include the version, exact rev and all the info in great detail about how you're doing this.

/b

On Mar 9, 2010, at 10:12 AM, Mark Sobkow wrote:

> I'm having a problem with call recording.  I tell Freeswitch to record a 
> call to a file, and it responds ok, and the file shows up on the file 
> system.  I later tell Freeswitch to stop recording, and it returns ok, 
> but the file keeps growing as it doesn't actually stop recording.
> 
> As a result, any subsequent attempts to start recording the session to a 
> different file name fails.
> 
> In my case, an operator is logged in to the phone system and receiving 
> calls from customers or from automated callbacks (if the customer didn't 
> want to wait.)  So I don't need to record the entire operator session, 
> just _segments_ of it.



From msc at freeswitch.org  Tue Mar  9 11:32:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Mar 2010 11:32:34 -0800
Subject: [Freeswitch-dev] mod lua session:bridged() not implemented?
In-Reply-To: <50e456911003081006g7ce1151fo285660906cdbaab7@mail.gmail.com>
References: <50e456911003081006g7ce1151fo285660906cdbaab7@mail.gmail.com>
Message-ID: <87f2f3b91003091132t3b57ec27u7d0f6d538535f8eb@mail.gmail.com>

On Mon, Mar 8, 2010 at 10:06 AM, rentmycoder rentmycoder <
rentmycoder at gmail.com> wrote:

> Yes you are absolutelly right, maybe I might be looking at the wrong
> tool...
> I have tried several approaches but haven't found the right one...
> I don't only need to detect if a call was answered, but I need also to
> detect and log the failure cause
> in case the call was not answered. Which usually can be found in the
> originate_disposition variable...
>
> If I use a socket command:
> "bgapi originate
> {ignore_early_media=true,originate_timeout=10,api_hangup_hook='luarun
> cc.lua predictive_hangup'}sofia/gateway/gwname/13246789 9999"
> I cannot detect the originate_disposition variable in
> predictive_hangup() in lua, becouse the session is nil after hangup...
>
> I think I should start dialing out from the dialplan, but in this case
> how to call to the dialplan using sockets?
> I've tried:
> "bgapi originate {phoneno=13246789}loopback/9999"
> with no success...
>
> What do you suggest?
> Is it possible to get this working only with 2 call legs???
>

It's been a while but I did this with two call legs a few years back. I was
using OpenZAP but the principles are the same: use the dialplan to process
the calls and if the call is answered you launch a script (or in my case,
transfer to a different dp extension) otherwise you wait for the call to
timeout.  It might be easier to discuss this in realtime on IRC, or you
could call the public conference tomorrow and after the main agenda and
Mathieu's mod_sofia presentation there will be an open floor for people to
ask questions and discuss FreeSWITCH topics.
-MC
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From msc at freeswitch.org  Tue Mar  9 22:50:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Mar 2010 22:50:59 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Agenda
Message-ID: <87f2f3b91003092250r6c32860ctde57e87563975f10@mail.gmail.com>

Hello all!

The FreeSWITCH weekly conference call agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_03_10

Please feel free to add your questions and agenda items. Also, remember that
this week Mathieu Rene is going to be spending a lot of time discussing
mod_sofia and how it works. It should be a very informative discussion! By
all means join us at 1700 GMT on Wednesday.

-Michael
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From james.gledhill at sipstorm.com  Wed Mar 10 09:23:42 2010
From: james.gledhill at sipstorm.com (Gledhill, James)
Date: Wed, 10 Mar 2010 12:23:42 -0500
Subject: [Freeswitch-dev] error events for failed calls using 'conference
	conf_name bgdial sofia/ip/user'
Message-ID: <A671FCA12875564E8CC11878981732BB0E17DFE2@mse2be2.mse2.exchange.ms>

 

My goals are:

1)       To get ?start-talking? and ?stop-talking? events during Early Media (before the call is answered) as well as after the call is answered.  

2)       If there are problems with the call, get events that tell me why the call failed.

 

The way I am doing it accomplishes the start/stop talking (in early media) BUT if the call fails, I do not get events back letting me know it failed and why.  I DO get CHANNEL_DESTROY and CHANNEL_STATE events that IF I had the UUID, I could match against, but at this point in the call I do NOT have that UUID AND when setting the UUID via ?origination_uuid=my_uuid_my_uuid_my_test_uuid?, my UUID has not yet been set, so as far as I can tell there is no way to match those events with details my application has at the time.  

 

The console shows the errors (Below are two examples of calls that failed in I entered ?bad? sofia locations), but they do not come back as events.

2010-03-05 14:52:20.109497 [ERR] switch_ivr_originate.c:2389 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED]

2010-03-05 16:34:13.669509 [ERR] switch_ivr_originate.c:2389 Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]

 

So ? I am wondering is there a better/different way to do it, is there a work-around, should I get events to tell me the dial had problems, etc.

 

I am using esl/lua and esl/java to experiment with this.  Both get the same results (as I would expect).

 

I have had this as a ~take-a-number on IRC for about a week, so I thought I would add more detail and open it up to this group.  I realize many of the same people do both, but hoped more eyes may look at it here and give thoughts.

 

 

===== From Pastebin ==== 

 

I am not getting 'error' events in ESL [that match a UUID I can track] when there is a problem with conference <conf_name> dial sofia/myip/1006, so I can not tell what went wrong.  Is there a work-around?  

 

 I start a call in a conference from ESL (conference conf_uuid-TEST_CONF bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.232/1006)

 

 If the 'dial' works (meaning does not get an error) then I get the events I want and expect.  If it errors (not able to make the call ( [USER_NOT_REGISTERED] | [DESTINATION_OUT_OF_ORDER] | etc. ) then I do not get any events that tell me something went wrong.

 

 Is there a work-around for this?

 

=====================

LUA Code  run from ./freeswitch.trunk/libs/esl/lua

 

#!/usr/local/bin/lua

require("ESL")

 

myip = "10.50.48.239"  -- change this to your IP address

myExt = 1006  -- change to one that is registered, then rerun with an unregistered

 

con = ESL.ESLconnection("localhost", "8021", "ClueCon");

 

str = string.format("conference conf_uuid-TEST_CONF bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/%s/%s", myip, myExt)

 

con:sendRecv("event plain ALL")

 

 

ff = con:api( str )

 

print("** sent " .. str )

 

print( (ff and ff:getBody() or "no body" ))

 

cnt = 0

while cnt <= 30 do

 

    print("loop = " .. cnt )

    cnt = cnt + 1

    ev = con:recvEventTimed(1000)

    print( ev and ev:serialize("plain") or "no event this time" )

 

end

 

==============

 

The printout is :

 

** sent conference conf_uuid-TEST_CONF bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.239/1006

OK Job-UUID: a5d57b34-9205-483c-ae64-60601d8178d7

 

loop = 0

Event-Name: CHANNEL_DESTROY

Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1

FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com

FreeSWITCH-IPv4: 10.50.48.232

FreeSWITCH-IPv6: %3A%3A1

Event-Date-Local: 2010-03-05%2017%3A06%3A52

Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT

Event-Date-Timestamp: 1267826812571550

Event-Calling-File: switch_core_session.c

Event-Calling-Function: switch_core_session_perform_destroy

Event-Calling-Line-Number: 1051

Channel-State: CS_NEW

Channel-State-Number: 0

Channel-Name: N/A

Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66

Call-Direction: outbound

Presence-Call-Direction: outbound

Answer-State: ringing

 

 

loop = 1

Event-Name: CHANNEL_STATE

Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1

FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com

FreeSWITCH-IPv4: 10.50.48.232

FreeSWITCH-IPv6: %3A%3A1

Event-Date-Local: 2010-03-05%2017%3A06%3A52

Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT

Event-Date-Timestamp: 1267826812571550

Event-Calling-File: switch_channel.c

Event-Calling-Function: switch_channel_perform_set_running_state

Event-Calling-Line-Number: 1301

Channel-State: CS_DESTROY

Channel-State-Number: 12

Channel-Name: _undef_

Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66

Call-Direction: outbound

Presence-Call-Direction: outbound

Answer-State: ringing

 

 

loop = 2

Event-Name: API

Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1

FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com

FreeSWITCH-IPv4: 10.50.48.232

FreeSWITCH-IPv6: %3A%3A1

Event-Date-Local: 2010-03-05%2017%3A06%3A52

Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT

Event-Date-Timestamp: 1267826812571550

Event-Calling-File: switch_loadable_module.c

Event-Calling-Function: switch_api_execute

Event-Calling-Line-Number: 1581

API-Command: conference

API-Command-Argument: conf_uuid-TEST_CONF%20bgdial%20%5Borigination_uuid%3Dmy_uuid_my_uuid_my_test_uuid%5Dsofia/10.50.48.239/1006%20

 

 

loop = 3

no event this time

loop = 4

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From brian at freeswitch.org  Wed Mar 10 10:04:52 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Mar 2010 12:04:52 -0600
Subject: [Freeswitch-dev] error events for failed calls using
	'conference conf_name bgdial sofia/ip/user'
In-Reply-To: <A671FCA12875564E8CC11878981732BB0E17DFE2@mse2be2.mse2.exchange.ms>
References: <A671FCA12875564E8CC11878981732BB0E17DFE2@mse2be2.mse2.exchange.ms>
Message-ID: <4709386B-A9F7-40CD-9F9A-A0705DA8AAD0@freeswitch.org>

http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings  

Please read that there is no such thing as sofia/ip/user  (unless the ip is an alias on the profile)

/b

On Mar 10, 2010, at 11:23 AM, Gledhill, James wrote:

>  
> My goals are:
> 1)       To get ?start-talking? and ?stop-talking? events during Early Media (before the call is answered) as well as after the call is answered. 
> 2)       If there are problems with the call, get events that tell me why the call failed.
>  
> The way I am doing it accomplishes the start/stop talking (in early media) BUT if the call fails, I do not get events back letting me know it failed and why.  I DO get CHANNEL_DESTROY and CHANNEL_STATE events that IF I had the UUID, I could match against, but at this point in the call I do NOT have that UUID AND when setting the UUID via ?origination_uuid=my_uuid_my_uuid_my_test_uuid?, my UUID has not yet been set, so as far as I can tell there is no way to match those events with details my application has at the time. 
>  
> The console shows the errors (Below are two examples of calls that failed in I entered ?bad? sofia locations), but they do not come back as events.
> 2010-03-05 14:52:20.109497 [ERR] switch_ivr_originate.c:2389 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED]
> 2010-03-05 16:34:13.669509 [ERR] switch_ivr_originate.c:2389 Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]
>  
> So ? I am wondering is there a better/different way to do it, is there a work-around, should I get events to tell me the dial had problems, etc.
>  
> I am using esl/lua and esl/java to experiment with this.  Both get the same results (as I would expect).
>  
> I have had this as a ~take-a-number on IRC for about a week, so I thought I would add more detail and open it up to this group.  I realize many of the same people do both, but hoped more eyes may look at it here and give thoughts.
>  
>  
> ===== From Pastebin ====
>  
> I am not getting 'error' events in ESL [that match a UUID I can track] when there is a problem with conference <conf_name> dial sofia/myip/1006, so I can not tell what went wrong.  Is there a work-around? 
>  
>  I start a call in a conference from ESL (conference conf_uuid-TEST_CONF bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.232/1006)
>  
>  If the 'dial' works (meaning does not get an error) then I get the events I want and expect.  If it errors (not able to make the call ( [USER_NOT_REGISTERED] | [DESTINATION_OUT_OF_ORDER] | etc. ) then I do not get any events that tell me something went wrong.
>  
>  Is there a work-around for this?
>  
> =====================
> LUA Code  run from ./freeswitch.trunk/libs/esl/lua
>  
> #!/usr/local/bin/lua
> require("ESL")
>  
> myip = "10.50.48.239"  -- change this to your IP address
> myExt = 1006  -- change to one that is registered, then rerun with an unregistered
>  
> con = ESL.ESLconnection("localhost", "8021", "ClueCon");
>  
> str = string.format("conference conf_uuid-TEST_CONF bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/%s/%s", myip, myExt)
>  
> con:sendRecv("event plain ALL")
>  
>  
> ff = con:api( str )
>  
> print("** sent " .. str )
>  
> print( (ff and ff:getBody() or "no body" ))
>  
> cnt = 0
> while cnt <= 30 do
>  
>     print("loop = " .. cnt )
>     cnt = cnt + 1
>     ev = con:recvEventTimed(1000)
>     print( ev and ev:serialize("plain") or "no event this time" )
>  
> end
>  
> ==============
>  
> The printout is :
>  
> ** sent conference conf_uuid-TEST_CONF bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.239/1006
> OK Job-UUID: a5d57b34-9205-483c-ae64-60601d8178d7
>  
> loop = 0
> Event-Name: CHANNEL_DESTROY
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
> FreeSWITCH-IPv4: 10.50.48.232
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
> Event-Date-Timestamp: 1267826812571550
> Event-Calling-File: switch_core_session.c
> Event-Calling-Function: switch_core_session_perform_destroy
> Event-Calling-Line-Number: 1051
> Channel-State: CS_NEW
> Channel-State-Number: 0
> Channel-Name: N/A
> Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66
> Call-Direction: outbound
> Presence-Call-Direction: outbound
> Answer-State: ringing
>  
>  
> loop = 1
> Event-Name: CHANNEL_STATE
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
> FreeSWITCH-IPv4: 10.50.48.232
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
> Event-Date-Timestamp: 1267826812571550
> Event-Calling-File: switch_channel.c
> Event-Calling-Function: switch_channel_perform_set_running_state
> Event-Calling-Line-Number: 1301
> Channel-State: CS_DESTROY
> Channel-State-Number: 12
> Channel-Name: _undef_
> Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66
> Call-Direction: outbound
> Presence-Call-Direction: outbound
> Answer-State: ringing
>  
>  
> loop = 2
> Event-Name: API
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
> FreeSWITCH-IPv4: 10.50.48.232
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
> Event-Date-Timestamp: 1267826812571550
> Event-Calling-File: switch_loadable_module.c
> Event-Calling-Function: switch_api_execute
> Event-Calling-Line-Number: 1581
> API-Command: conference
> API-Command-Argument: conf_uuid-TEST_CONF%20bgdial%20%5Borigination_uuid%3Dmy_uuid_my_uuid_my_test_uuid%5Dsofia/10.50.48.239/1006%20
>  
>  
> loop = 3
> no event this time
> loop = 4
> _______________________________________________
> 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 Mar 10 10:15:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 12:15:30 -0600
Subject: [Freeswitch-dev] error events for failed calls using
	'conference conf_name bgdial sofia/ip/user'
In-Reply-To: <4709386B-A9F7-40CD-9F9A-A0705DA8AAD0@freeswitch.org>
References: <A671FCA12875564E8CC11878981732BB0E17DFE2@mse2be2.mse2.exchange.ms>
	<4709386B-A9F7-40CD-9F9A-A0705DA8AAD0@freeswitch.org>
Message-ID: <191c3a031003101015i17f350fanccbe6e8d2f60f7d0@mail.gmail.com>

consider

api originate <dialstring> conference conf_uuid-TEST_CONF inline

this returns success+uuid or fail


On Wed, Mar 10, 2010 at 12:04 PM, Brian West <brian at freeswitch.org> wrote:

> http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings
>
> Please read that there is no such thing as sofia/ip/user  (unless the ip is
> an alias on the profile)
>
> /b
>
> On Mar 10, 2010, at 11:23 AM, Gledhill, James wrote:
>
>
>
> My goals are:
>
> 1)       To get ?start-talking? and ?stop-talking? events during Early
> Media (before the call is answered) as well as after the call is answered.
>
> 2)       If there are problems with the call, get events that tell me why
> the call failed.
>
>
>
> The way I am doing it accomplishes the start/stop talking (in early media)
> BUT if the call fails, I do not get events back letting me know it failed
> and why.  I DO get CHANNEL_DESTROY and CHANNEL_STATE events that IF I had
> the UUID, I could match against, but at this point in the call I do NOT have
> that UUID AND when setting the UUID via
> ?origination_uuid=my_uuid_my_uuid_my_test_uuid?, my UUID has not yet been
> set, so as far as I can tell there is no way to match those events with
> details my application has at the time.
>
>
>
> The console shows the errors (Below are two examples of calls that failed
> in I entered ?bad? sofia locations), but they do not come back as events.
>
> 2010-03-05 14:52:20.109497 [ERR] switch_ivr_originate.c:2389 Cannot create
> outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED]
>
> 2010-03-05 16:34:13.669509 [ERR] switch_ivr_originate.c:2389 Cannot create
> outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]
>
>
>
> So ? I am wondering is there a better/different way to do it, is there a
> work-around, should I get events to tell me the dial had problems, etc.
>
>
>
> I am using esl/lua and esl/java to experiment with this.  Both get the same
> results (as I would expect).
>
>
>
> I have had this as a ~take-a-number on IRC for about a week, so I thought I
> would add more detail and open it up to this group.  I realize many of the
> same people do both, but hoped more eyes may look at it here and give
> thoughts.
>
>
>
>
>
> ===== From Pastebin ====
>
>
>
> I am not getting 'error' events in ESL [that match a UUID I can track] when
> there is a problem with conference <conf_name> dial sofia/myip/1006, so I
> can not tell what went wrong.  Is there a work-around?
>
>
>
>  I start a call in a conference from ESL (conference conf_uuid-TEST_CONF
> bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/
> 10.50.48.232/1006)
>
>
>
>  If the 'dial' works (meaning does not get an error) then I get the events
> I want and expect.  If it errors (not able to make the call (
> [USER_NOT_REGISTERED] | [DESTINATION_OUT_OF_ORDER] | etc. ) then I do not
> get any events that tell me something went wrong.
>
>
>
>  Is there a work-around for this?
>
>
>
> =====================
>
> LUA Code  run from ./freeswitch.trunk/libs/esl/lua
>
>
>
> #!/usr/local/bin/lua
>
> require("ESL")
>
>
>
> myip = "10.50.48.239"  -- change this to your IP address
>
> myExt = 1006  -- change to one that is registered, then rerun with an
> unregistered
>
>
>
> con = ESL.ESLconnection("localhost", "8021", "ClueCon");
>
>
>
> str = string.format("conference conf_uuid-TEST_CONF bgdial
> [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/%s/%s", myip, myExt)
>
>
>
> con:sendRecv("event plain ALL")
>
>
>
>
>
> ff = con:api( str )
>
>
>
> print("** sent " .. str )
>
>
>
> print( (ff and ff:getBody() or "no body" ))
>
>
>
> cnt = 0
>
> while cnt <= 30 do
>
>
>
>     print("loop = " .. cnt )
>
>     cnt = cnt + 1
>
>     ev = con:recvEventTimed(1000)
>
>     print( ev and ev:serialize("plain") or "no event this time" )
>
>
>
> end
>
>
>
> ==============
>
>
>
> The printout is :
>
>
>
> ** sent conference conf_uuid-TEST_CONF bgdial
> [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.239/1006
>
> OK Job-UUID: a5d57b34-9205-483c-ae64-60601d8178d7
>
>
>
> loop = 0
>
> Event-Name: CHANNEL_DESTROY
>
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
>
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
>
> FreeSWITCH-IPv4: 10.50.48.232
>
> FreeSWITCH-IPv6: %3A%3A1
>
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
>
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
>
> Event-Date-Timestamp: 1267826812571550
>
> Event-Calling-File: switch_core_session.c
>
> Event-Calling-Function: switch_core_session_perform_destroy
>
> Event-Calling-Line-Number: 1051
>
> Channel-State: CS_NEW
>
> Channel-State-Number: 0
>
> Channel-Name: N/A
>
> Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66
>
> Call-Direction: outbound
>
> Presence-Call-Direction: outbound
>
> Answer-State: ringing
>
>
>
>
>
> loop = 1
>
> Event-Name: CHANNEL_STATE
>
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
>
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
>
> FreeSWITCH-IPv4: 10.50.48.232
>
> FreeSWITCH-IPv6: %3A%3A1
>
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
>
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
>
> Event-Date-Timestamp: 1267826812571550
>
> Event-Calling-File: switch_channel.c
>
> Event-Calling-Function: switch_channel_perform_set_running_state
>
> Event-Calling-Line-Number: 1301
>
> Channel-State: CS_DESTROY
>
> Channel-State-Number: 12
>
> Channel-Name: _undef_
>
> Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66
>
> Call-Direction: outbound
>
> Presence-Call-Direction: outbound
>
> Answer-State: ringing
>
>
>
>
>
> loop = 2
>
> Event-Name: API
>
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
>
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
>
> FreeSWITCH-IPv4: 10.50.48.232
>
> FreeSWITCH-IPv6: %3A%3A1
>
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
>
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
>
> Event-Date-Timestamp: 1267826812571550
>
> Event-Calling-File: switch_loadable_module.c
>
> Event-Calling-Function: switch_api_execute
>
> Event-Calling-Line-Number: 1581
>
> API-Command: conference
>
> API-Command-Argument:
> conf_uuid-TEST_CONF%20bgdial%20%5Borigination_uuid%3Dmy_uuid_my_uuid_my_test_uuid%5Dsofia/
> 10.50.48.239/1006%20
>
>
>
>
>
> loop = 3
>
> no event this time
>
> loop = 4
>  _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From james.gledhill at sipstorm.com  Wed Mar 10 11:29:43 2010
From: james.gledhill at sipstorm.com (Gledhill, James)
Date: Wed, 10 Mar 2010 14:29:43 -0500
Subject: [Freeswitch-dev] error events for failed calls using'conference
	conf_name bgdial sofia/ip/user'
In-Reply-To: <191c3a031003101015i17f350fanccbe6e8d2f60f7d0@mail.gmail.com>
Message-ID: <A671FCA12875564E8CC11878981732BB0E17E0AE@mse2be2.mse2.exchange.ms>

Anthony - 

 

The problem with api originate <dialstring> conference
conf_uuid-TEST_CONF inline is that it does not return until after the
call is answered - so I do not have events during early media.  I did
try to add {ignore_early_media=true} but it still did not return until
after I answered the phone (using esl lua for this test)

 

I also tried "bgapi originate
{ignore_early_media=true,origination_uuid=my_uuid_my_uuid_my_test_uuid}s
ofia/internal/+18015551212 at 10.50.40.50:5060
&conference(conf_uuid-TEST_CONF) inline", (I used a different TN, but
changed it for privacy) which makes the call and returns events before I
answer the call, BUT the conf bridge does not get involved (called)
until AFTER the origination call is answered.

 

Neither of these allows me to get "start-talking" and "stop-talking"
events during Early Media from the conf bridge because the conference is
not involved during the early media part of the call.

 

Brian - 

 

I should have been more detailed in my sofia string in the subject.  I
am aware that sofia/ip/user is not valid. Two examples sofia string I
use are below.

* If I used sofia/10.50.48.222/1002 AND that user is registered and
answers, then all works well.  If that user is not registered, I need to
get USER_NOT_REGISTERED

* If I use sofia/internal/+18015 at 10.50.40.50 (which is not a valid E.164
TN) then I should get DESTINATION_OUT_OF_ORDER or some other error
message.

 

 

________________________________

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Wednesday, March 10, 2010 12:16 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] error events for failed calls
using'conference conf_name bgdial sofia/ip/user'

 

consider 

api originate <dialstring> conference conf_uuid-TEST_CONF inline

this returns success+uuid or fail



On Wed, Mar 10, 2010 at 12:04 PM, Brian West <brian at freeswitch.org>
wrote:

http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings  

 

Please read that there is no such thing as sofia/ip/user  (unless the ip
is an alias on the profile)

 

/b

 

On Mar 10, 2010, at 11:23 AM, Gledhill, James wrote:

 

	 

	My goals are:

	1)       To get "start-talking" and "stop-talking" events during
Early Media (before the call is answered) as well as after the call is
answered.  

	2)       If there are problems with the call, get events that
tell me why the call failed.

	 

	The way I am doing it accomplishes the start/stop talking (in
early media) BUT if the call fails, I do not get events back letting me
know it failed and why.  I DO get CHANNEL_DESTROY and CHANNEL_STATE
events that IF I had the UUID, I could match against, but at this point
in the call I do NOT have that UUID AND when setting the UUID via
"origination_uuid=my_uuid_my_uuid_my_test_uuid", my UUID has not yet
been set, so as far as I can tell there is no way to match those events
with details my application has at the time.  

	 

	The console shows the errors (Below are two examples of calls
that failed in I entered "bad" sofia locations), but they do not come
back as events.

	2010-03-05 14:52:20.109497 [ERR] switch_ivr_originate.c:2389
Cannot create outgoing channel of type [sofia] cause:
[USER_NOT_REGISTERED]

	2010-03-05 16:34:13.669509 [ERR] switch_ivr_originate.c:2389
Cannot create outgoing channel of type [sofia] cause:
[DESTINATION_OUT_OF_ORDER]

	 

	So ... I am wondering is there a better/different way to do it,
is there a work-around, should I get events to tell me the dial had
problems, etc.

	 

	I am using esl/lua and esl/java to experiment with this.  Both
get the same results (as I would expect).

	 

	I have had this as a ~take-a-number on IRC for about a week, so
I thought I would add more detail and open it up to this group.  I
realize many of the same people do both, but hoped more eyes may look at
it here and give thoughts.

	 

	 

	===== From Pastebin ==== 

	 

	I am not getting 'error' events in ESL [that match a UUID I can
track] when there is a problem with conference <conf_name> dial
sofia/myip/1006, so I can not tell what went wrong.  Is there a
work-around?  

	 

	 I start a call in a conference from ESL (conference
conf_uuid-TEST_CONF bgdial
[origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.232/1006)

	 

	 If the 'dial' works (meaning does not get an error) then I get
the events I want and expect.  If it errors (not able to make the call (
[USER_NOT_REGISTERED] | [DESTINATION_OUT_OF_ORDER] | etc. ) then I do
not get any events that tell me something went wrong.

	 

	 Is there a work-around for this?

	 

	=====================

	LUA Code  run from ./freeswitch.trunk/libs/esl/lua

	 

	#!/usr/local/bin/lua

	require("ESL")

	 

	myip = "10.50.48.239"  -- change this to your IP address

	myExt = 1006  -- change to one that is registered, then rerun
with an unregistered

	 

	con = ESL.ESLconnection("localhost", "8021", "ClueCon");

	 

	str = string.format("conference conf_uuid-TEST_CONF bgdial
[origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/%s/%s", myip,
myExt)

	 

	con:sendRecv("event plain ALL")

	 

	 

	ff = con:api( str )

	 

	print("** sent " .. str )

	 

	print( (ff and ff:getBody() or "no body" ))

	 

	cnt = 0

	while cnt <= 30 do

	 

	    print("loop = " .. cnt )

	    cnt = cnt + 1

	    ev = con:recvEventTimed(1000)

	    print( ev and ev:serialize("plain") or "no event this time"
)

	 

	end

	 

	==============

	 

	The printout is :

	 

	** sent conference conf_uuid-TEST_CONF bgdial
[origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.239/1006

	OK Job-UUID: a5d57b34-9205-483c-ae64-60601d8178d7

	 

	loop = 0

	Event-Name: CHANNEL_DESTROY

	Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1

	FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com

	FreeSWITCH-IPv4: 10.50.48.232

	FreeSWITCH-IPv6: %3A%3A1

	Event-Date-Local: 2010-03-05%2017%3A06%3A52

	Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT

	Event-Date-Timestamp: 1267826812571550

	Event-Calling-File: switch_core_session.c

	Event-Calling-Function: switch_core_session_perform_destroy

	Event-Calling-Line-Number: 1051

	Channel-State: CS_NEW

	Channel-State-Number: 0

	Channel-Name: N/A

	Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66

	Call-Direction: outbound

	Presence-Call-Direction: outbound

	Answer-State: ringing

	 

	 

	loop = 1

	Event-Name: CHANNEL_STATE

	Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1

	FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com

	FreeSWITCH-IPv4: 10.50.48.232

	FreeSWITCH-IPv6: %3A%3A1

	Event-Date-Local: 2010-03-05%2017%3A06%3A52

	Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT

	Event-Date-Timestamp: 1267826812571550

	Event-Calling-File: switch_channel.c

	Event-Calling-Function: switch_channel_perform_set_running_state

	Event-Calling-Line-Number: 1301

	Channel-State: CS_DESTROY

	Channel-State-Number: 12

	Channel-Name: _undef_

	Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66

	Call-Direction: outbound

	Presence-Call-Direction: outbound

	Answer-State: ringing

	 

	 

	loop = 2

	Event-Name: API

	Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1

	FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com

	FreeSWITCH-IPv4: 10.50.48.232

	FreeSWITCH-IPv6: %3A%3A1

	Event-Date-Local: 2010-03-05%2017%3A06%3A52

	Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT

	Event-Date-Timestamp: 1267826812571550

	Event-Calling-File: switch_loadable_module.c

	Event-Calling-Function: switch_api_execute

	Event-Calling-Line-Number: 1581

	API-Command: conference

	API-Command-Argument:
conf_uuid-TEST_CONF%20bgdial%20%5Borigination_uuid%3Dmy_uuid_my_uuid_my_
test_uuid%5Dsofia/10.50.48.239/1006%20

	 

	 

	loop = 3

	no event this time

	loop = 4

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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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From anthony.minessale at gmail.com  Wed Mar 10 12:35:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 14:35:50 -0600
Subject: [Freeswitch-dev] error events for failed calls using'conference
	conf_name bgdial sofia/ip/user'
In-Reply-To: <A671FCA12875564E8CC11878981732BB0E17E0AE@mse2be2.mse2.exchange.ms>
References: <191c3a031003101015i17f350fanccbe6e8d2f60f7d0@mail.gmail.com>
	<A671FCA12875564E8CC11878981732BB0E17E0AE@mse2be2.mse2.exchange.ms>
Message-ID: <191c3a031003101235x4f7ad90dn6dd141d5a4dcaedb@mail.gmail.com>

Some of your findings are incorrect.

The default behavior of originate is to return as soon as there is early
media. (sip 183)
Adding ignore_early_media=true is the opposite effect so you do not want
that.

bgapi just does the command in a separate thread so you don't want that
either.

*api originate <dialstring> conference conf_uuid-TEST_CONF inline*
will put the call in the conference the very instant it sees a 183

If you see anything other than this, make sure you are on latest trunk and
do a console trace demonstrating otherwise.

use these commands for more details.

console loglevel debug
sofia profile internal siptrace on



On Wed, Mar 10, 2010 at 1:29 PM, Gledhill, James <
james.gledhill at sipstorm.com> wrote:

>  Anthony ?
>
>
>
> The problem with *api originate <dialstring> conference
> conf_uuid-TEST_CONF inline* is that it does not return until after the
> call is answered ? so I do not have events during early media.  I did try to
> add *{ignore_early_media=true}* but it still did not return until after I
> answered the phone (using esl lua for this test)
>
>
>
> I also tried "*bgapi originate
> {ignore_early_media=true,origination_uuid=my_uuid_my_uuid_my_test_uuid}sofia/internal/+
> 18015551212 at 10.50.40.50:5060 &conference(conf_uuid-TEST_CONF) inline*?, (I
> used a different TN, but changed it for privacy) which makes the call and
> returns events before I answer the call, BUT the conf bridge does not get
> involved (called) until AFTER the origination call is answered.
>
>
>
> Neither of these allows me to get ?start-talking? and ?stop-talking? events
> during Early Media from the conf bridge because the conference is not
> involved during the early media part of the call.
>
>
>
> Brian ?
>
>
>
> I should have been more detailed in my sofia string in the subject.  I am
> aware that sofia/ip/user is not valid. Two examples sofia string I use are
> below.**
>
> * If I used sofia/10.50.48.222/1002 AND that user is registered and
> answers, then all works well.  If that user is not registered, I need to get
> USER_NOT_REGISTERED
>
> * If I use sofia/internal/+18015 at 10.50.40.50 (which is not a valid E.164
> TN) then I should get DESTINATION_OUT_OF_ORDER or some other error message.
>
>
>
>
>  ------------------------------
>
> *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, March 10, 2010 12:16 PM
> *To:* freeswitch-dev at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-dev] error events for failed calls
> using'conference conf_name bgdial sofia/ip/user'
>
>
>
> consider
>
> api originate <dialstring> conference conf_uuid-TEST_CONF inline
>
> this returns success+uuid or fail
>
>  On Wed, Mar 10, 2010 at 12:04 PM, Brian West <brian at freeswitch.org>
> wrote:
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings
>
>
>
> Please read that there is no such thing as sofia/ip/user  (unless the ip is
> an alias on the profile)
>
>
>
> /b
>
>
>
> On Mar 10, 2010, at 11:23 AM, Gledhill, James wrote:
>
>
>
>
>
> My goals are:
>
> 1)       To get ?start-talking? and ?stop-talking? events during Early
> Media (before the call is answered) as well as after the call is answered.
>
> 2)       If there are problems with the call, get events that tell me why
> the call failed.
>
>
>
> The way I am doing it accomplishes the start/stop talking (in early media)
> BUT if the call fails, I do not get events back letting me know it failed
> and why.  I DO get CHANNEL_DESTROY and CHANNEL_STATE events that IF I had
> the UUID, I could match against, but at this point in the call I do NOT have
> that UUID AND when setting the UUID via
> ?origination_uuid=my_uuid_my_uuid_my_test_uuid?, my UUID has not yet been
> set, so as far as I can tell there is no way to match those events with
> details my application has at the time.
>
>
>
> The console shows the errors (Below are two examples of calls that failed
> in I entered ?bad? sofia locations), but they do not come back as events.
>
> 2010-03-05 14:52:20.109497 [ERR] switch_ivr_originate.c:2389 Cannot create
> outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED]
>
> 2010-03-05 16:34:13.669509 [ERR] switch_ivr_originate.c:2389 Cannot create
> outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]
>
>
>
> So ? I am wondering is there a better/different way to do it, is there a
> work-around, should I get events to tell me the dial had problems, etc.
>
>
>
> I am using esl/lua and esl/java to experiment with this.  Both get the same
> results (as I would expect).
>
>
>
> I have had this as a ~take-a-number on IRC for about a week, so I thought I
> would add more detail and open it up to this group.  I realize many of the
> same people do both, but hoped more eyes may look at it here and give
> thoughts.
>
>
>
>
>
> ===== From Pastebin ====
>
>
>
> I am not getting 'error' events in ESL [that match a UUID I can track] when
> there is a problem with conference <conf_name> dial sofia/myip/1006, so I
> can not tell what went wrong.  Is there a work-around?
>
>
>
>  I start a call in a conference from ESL (conference conf_uuid-TEST_CONF
> bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/
> 10.50.48.232/1006)
>
>
>
>  If the 'dial' works (meaning does not get an error) then I get the events
> I want and expect.  If it errors (not able to make the call (
> [USER_NOT_REGISTERED] | [DESTINATION_OUT_OF_ORDER] | etc. ) then I do not
> get any events that tell me something went wrong.
>
>
>
>  Is there a work-around for this?
>
>
>
> =====================
>
> LUA Code  run from ./freeswitch.trunk/libs/esl/lua
>
>
>
> #!/usr/local/bin/lua
>
> require("ESL")
>
>
>
> myip = "10.50.48.239"  -- change this to your IP address
>
> myExt = 1006  -- change to one that is registered, then rerun with an
> unregistered
>
>
>
> con = ESL.ESLconnection("localhost", "8021", "ClueCon");
>
>
>
> str = string.format("conference conf_uuid-TEST_CONF bgdial
> [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/%s/%s", myip, myExt)
>
>
>
> con:sendRecv("event plain ALL")
>
>
>
>
>
> ff = con:api( str )
>
>
>
> print("** sent " .. str )
>
>
>
> print( (ff and ff:getBody() or "no body" ))
>
>
>
> cnt = 0
>
> while cnt <= 30 do
>
>
>
>     print("loop = " .. cnt )
>
>     cnt = cnt + 1
>
>     ev = con:recvEventTimed(1000)
>
>     print( ev and ev:serialize("plain") or "no event this time" )
>
>
>
> end
>
>
>
> ==============
>
>
>
> The printout is :
>
>
>
> ** sent conference conf_uuid-TEST_CONF bgdial
> [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.239/1006
>
> OK Job-UUID: a5d57b34-9205-483c-ae64-60601d8178d7
>
>
>
> loop = 0
>
> Event-Name: CHANNEL_DESTROY
>
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
>
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
>
> FreeSWITCH-IPv4: 10.50.48.232
>
> FreeSWITCH-IPv6: %3A%3A1
>
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
>
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
>
> Event-Date-Timestamp: 1267826812571550
>
> Event-Calling-File: switch_core_session.c
>
> Event-Calling-Function: switch_core_session_perform_destroy
>
> Event-Calling-Line-Number: 1051
>
> Channel-State: CS_NEW
>
> Channel-State-Number: 0
>
> Channel-Name: N/A
>
> Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66
>
> Call-Direction: outbound
>
> Presence-Call-Direction: outbound
>
> Answer-State: ringing
>
>
>
>
>
> loop = 1
>
> Event-Name: CHANNEL_STATE
>
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
>
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
>
> FreeSWITCH-IPv4: 10.50.48.232
>
> FreeSWITCH-IPv6: %3A%3A1
>
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
>
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
>
> Event-Date-Timestamp: 1267826812571550
>
> Event-Calling-File: switch_channel.c
>
> Event-Calling-Function: switch_channel_perform_set_running_state
>
> Event-Calling-Line-Number: 1301
>
> Channel-State: CS_DESTROY
>
> Channel-State-Number: 12
>
> Channel-Name: _undef_
>
> Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66
>
> Call-Direction: outbound
>
> Presence-Call-Direction: outbound
>
> Answer-State: ringing
>
>
>
>
>
> loop = 2
>
> Event-Name: API
>
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
>
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
>
> FreeSWITCH-IPv4: 10.50.48.232
>
> FreeSWITCH-IPv6: %3A%3A1
>
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
>
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
>
> Event-Date-Timestamp: 1267826812571550
>
> Event-Calling-File: switch_loadable_module.c
>
> Event-Calling-Function: switch_api_execute
>
> Event-Calling-Line-Number: 1581
>
> API-Command: conference
>
> API-Command-Argument:
> conf_uuid-TEST_CONF%20bgdial%20%5Borigination_uuid%3Dmy_uuid_my_uuid_my_test_uuid%5Dsofia/
> 10.50.48.239/1006%20
>
>
>
>
>
> loop = 3
>
> no event this time
>
> loop = 4
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+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
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From james.gledhill at sipstorm.com  Wed Mar 10 14:38:38 2010
From: james.gledhill at sipstorm.com (Gledhill, James)
Date: Wed, 10 Mar 2010 17:38:38 -0500
Subject: [Freeswitch-dev] error events for failed calls
	using'conferenceconf_name bgdial sofia/ip/user'
In-Reply-To: <191c3a031003101235x4f7ad90dn6dd141d5a4dcaedb@mail.gmail.com>
Message-ID: <A671FCA12875564E8CC11878981732BB0E17E289@mse2be2.mse2.exchange.ms>

Thanks Anthony - 

 

I got it to work with the following:

 

FS Version : 16852 and later 16963M

 

"api originate sofia/10.50.48.232/+{my_tn_here}@10.50.50.50:5060
&conference(conf_uuid-TEST_CON) inline"

 

If I used api originate <dialstring> conference conf_uuid-TEST_CONF
inline then FS tried to make a conference of conf_uuid-TEST_CON at inline
.. using &conference(conf_uuid-TEST_CON) inline fixed that

 

2010-03-10 17:03:36.463558 [NOTICE] switch_ivr.c:1447 Transfer
sofia/internal/+{my_tn}@{my_gw} to conf_uuid-TEST_CON[conference at inline]

2010-03-10 17:03:36.465638 [DEBUG] switch_core_state_machine.c:133 No
Dialplan, changing state to CONSUME_MEDIA

 

 

---------

 

For others that may look at this later, my test code to get
"start-talking" and "stop-talking" events during Early Media using esl
lua (freesiwtch_src/libs/esl/lua) is

 

 

#!/usr/local/bin/lua

require("ESL")

 

local con = ESL.ESLconnection("localhost", "8021", "ClueCon");

 

con:sendRecv("event plain ALL")

ob_session_string = string.format("originate
sofia/internal/{MY_TN_NUMBER_HERE}@{MY_GATEWAY_HERE}
&conference(conf_uuid-TEST_CON) inline")

 

ff = con:api( ob_session_string )

print("** sent api")

 

if ff then

    print(ff:getBody())

    print(ff: getHeader("result"))

else

    print("no body")

end

 

 

cnt = 0

while cnt <= 40 do

 

    print("loop = " .. cnt )

    cnt = cnt + 1

    ev = con:recvEventTimed(1000)

    print( ev and ev:serialize("plain") or "no event this time" )

 

end

________________________________

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Wednesday, March 10, 2010 2:36 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] error events for failed calls
using'conferenceconf_name bgdial sofia/ip/user'

 

Some of your findings are incorrect.

The default behavior of originate is to return as soon as there is early
media. (sip 183)
Adding ignore_early_media=true is the opposite effect so you do not want
that.

bgapi just does the command in a separate thread so you don't want that
either.

api originate <dialstring> conference conf_uuid-TEST_CONF inline
will put the call in the conference the very instant it sees a 183 

If you see anything other than this, make sure you are on latest trunk
and do a console trace demonstrating otherwise.

use these commands for more details.

console loglevel debug
sofia profile internal siptrace on




On Wed, Mar 10, 2010 at 1:29 PM, Gledhill, James
<james.gledhill at sipstorm.com> wrote:

Anthony - 

 

The problem with api originate <dialstring> conference
conf_uuid-TEST_CONF inline is that it does not return until after the
call is answered - so I do not have events during early media.  I did
try to add {ignore_early_media=true} but it still did not return until
after I answered the phone (using esl lua for this test)

 

I also tried "bgapi originate
{ignore_early_media=true,origination_uuid=my_uuid_my_uuid_my_test_uuid}s
ofia/internal/+18015551212 at 10.50.40.50:5060
&conference(conf_uuid-TEST_CONF) inline", (I used a different TN, but
changed it for privacy) which makes the call and returns events before I
answer the call, BUT the conf bridge does not get involved (called)
until AFTER the origination call is answered.

 

Neither of these allows me to get "start-talking" and "stop-talking"
events during Early Media from the conf bridge because the conference is
not involved during the early media part of the call.

 

Brian - 

 

I should have been more detailed in my sofia string in the subject.  I
am aware that sofia/ip/user is not valid. Two examples sofia string I
use are below.

* If I used sofia/10.50.48.222/1002 AND that user is registered and
answers, then all works well.  If that user is not registered, I need to
get USER_NOT_REGISTERED

* If I use sofia/internal/+18015 at 10.50.40.50 (which is not a valid E.164
TN) then I should get DESTINATION_OUT_OF_ORDER or some other error
message.

 

 

________________________________

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Wednesday, March 10, 2010 12:16 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] error events for failed calls
using'conference conf_name bgdial sofia/ip/user'

 

consider 

api originate <dialstring> conference conf_uuid-TEST_CONF inline

this returns success+uuid or fail

On Wed, Mar 10, 2010 at 12:04 PM, Brian West <brian at freeswitch.org>
wrote:

http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings  

 

Please read that there is no such thing as sofia/ip/user  (unless the ip
is an alias on the profile)

 

/b

 

On Mar 10, 2010, at 11:23 AM, Gledhill, James wrote:

 

	 

	My goals are:

	1)       To get "start-talking" and "stop-talking" events during
Early Media (before the call is answered) as well as after the call is
answered.  

	2)       If there are problems with the call, get events that
tell me why the call failed.

	 

	The way I am doing it accomplishes the start/stop talking (in
early media) BUT if the call fails, I do not get events back letting me
know it failed and why.  I DO get CHANNEL_DESTROY and CHANNEL_STATE
events that IF I had the UUID, I could match against, but at this point
in the call I do NOT have that UUID AND when setting the UUID via
"origination_uuid=my_uuid_my_uuid_my_test_uuid", my UUID has not yet
been set, so as far as I can tell there is no way to match those events
with details my application has at the time.  

	 

	The console shows the errors (Below are two examples of calls
that failed in I entered "bad" sofia locations), but they do not come
back as events.

	2010-03-05 14:52:20.109497 [ERR] switch_ivr_originate.c:2389
Cannot create outgoing channel of type [sofia] cause:
[USER_NOT_REGISTERED]

	2010-03-05 16:34:13.669509 [ERR] switch_ivr_originate.c:2389
Cannot create outgoing channel of type [sofia] cause:
[DESTINATION_OUT_OF_ORDER]

	 

	So ... I am wondering is there a better/different way to do it,
is there a work-around, should I get events to tell me the dial had
problems, etc.

	 

	I am using esl/lua and esl/java to experiment with this.  Both
get the same results (as I would expect).

	 

	I have had this as a ~take-a-number on IRC for about a week, so
I thought I would add more detail and open it up to this group.  I
realize many of the same people do both, but hoped more eyes may look at
it here and give thoughts.

	 

	 

	===== From Pastebin ==== 

	 

	I am not getting 'error' events in ESL [that match a UUID I can
track] when there is a problem with conference <conf_name> dial
sofia/myip/1006, so I can not tell what went wrong.  Is there a
work-around?  

	 

	 I start a call in a conference from ESL (conference
conf_uuid-TEST_CONF bgdial
[origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.232/1006)

	 

	 If the 'dial' works (meaning does not get an error) then I get
the events I want and expect.  If it errors (not able to make the call (
[USER_NOT_REGISTERED] | [DESTINATION_OUT_OF_ORDER] | etc. ) then I do
not get any events that tell me something went wrong.

	 

	 Is there a work-around for this?

	 

	=====================

	LUA Code  run from ./freeswitch.trunk/libs/esl/lua

	 

	#!/usr/local/bin/lua

	require("ESL")

	 

	myip = "10.50.48.239"  -- change this to your IP address

	myExt = 1006  -- change to one that is registered, then rerun
with an unregistered

	 

	con = ESL.ESLconnection("localhost", "8021", "ClueCon");

	 

	str = string.format("conference conf_uuid-TEST_CONF bgdial
[origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/%s/%s", myip,
myExt)

	 

	con:sendRecv("event plain ALL")

	 

	 

	ff = con:api( str )

	 

	print("** sent " .. str )

	 

	print( (ff and ff:getBody() or "no body" ))

	 

	cnt = 0

	while cnt <= 30 do

	 

	    print("loop = " .. cnt )

	    cnt = cnt + 1

	    ev = con:recvEventTimed(1000)

	    print( ev and ev:serialize("plain") or "no event this time"
)

	 

	end

	 

	==============

	 

	The printout is :

	 

	** sent conference conf_uuid-TEST_CONF bgdial
[origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.239/1006

	OK Job-UUID: a5d57b34-9205-483c-ae64-60601d8178d7

	 

	loop = 0

	Event-Name: CHANNEL_DESTROY

	Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1

	FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com

	FreeSWITCH-IPv4: 10.50.48.232

	FreeSWITCH-IPv6: %3A%3A1

	Event-Date-Local: 2010-03-05%2017%3A06%3A52

	Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT

	Event-Date-Timestamp: 1267826812571550

	Event-Calling-File: switch_core_session.c

	Event-Calling-Function: switch_core_session_perform_destroy

	Event-Calling-Line-Number: 1051

	Channel-State: CS_NEW

	Channel-State-Number: 0

	Channel-Name: N/A

	Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66

	Call-Direction: outbound

	Presence-Call-Direction: outbound

	Answer-State: ringing

	 

	 

	loop = 1

	Event-Name: CHANNEL_STATE

	Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1

	FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com

	FreeSWITCH-IPv4: 10.50.48.232

	FreeSWITCH-IPv6: %3A%3A1

	Event-Date-Local: 2010-03-05%2017%3A06%3A52

	Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT

	Event-Date-Timestamp: 1267826812571550

	Event-Calling-File: switch_channel.c

	Event-Calling-Function: switch_channel_perform_set_running_state

	Event-Calling-Line-Number: 1301

	Channel-State: CS_DESTROY

	Channel-State-Number: 12

	Channel-Name: _undef_

	Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66

	Call-Direction: outbound

	Presence-Call-Direction: outbound

	Answer-State: ringing

	 

	 

	loop = 2

	Event-Name: API

	Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1

	FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com

	FreeSWITCH-IPv4: 10.50.48.232

	FreeSWITCH-IPv6: %3A%3A1

	Event-Date-Local: 2010-03-05%2017%3A06%3A52

	Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT

	Event-Date-Timestamp: 1267826812571550

	Event-Calling-File: switch_loadable_module.c

	Event-Calling-Function: switch_api_execute

	Event-Calling-Line-Number: 1581

	API-Command: conference

	API-Command-Argument:
conf_uuid-TEST_CONF%20bgdial%20%5Borigination_uuid%3Dmy_uuid_my_uuid_my_
test_uuid%5Dsofia/10.50.48.239/1006%20

	 

	 

	loop = 3

	no event this time

	loop = 4

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-- 
Anthony Minessale II

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

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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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FreeSWITCH Developer Conference
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<mailto:sip%3A888 at conference.freeswitch.org> 
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From anthony.minessale at gmail.com  Wed Mar 10 14:59:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 16:59:17 -0600
Subject: [Freeswitch-dev] error events for failed calls
	using'conferenceconf_name bgdial sofia/ip/user'
In-Reply-To: <A671FCA12875564E8CC11878981732BB0E17E289@mse2be2.mse2.exchange.ms>
References: <191c3a031003101235x4f7ad90dn6dd141d5a4dcaedb@mail.gmail.com>
	<A671FCA12875564E8CC11878981732BB0E17E289@mse2be2.mse2.exchange.ms>
Message-ID: <191c3a031003101459h19317747y8081d0f97e91b54b@mail.gmail.com>

&app(arg) removes the requirement for "inline"

inline is a special dialplan that uses a list of apps as the extension input

I think I made a typo in my example

*api originate <dialstring> conference:conf_uuid-TEST_CONF inline*

the app args should be connected to the app name with a : char when using
the inline dp

The 2 inputs to originate following the dialstring are

&app(args) OR <extension> <dialplan> <context>


so conference:foo inline is equiv to &conference(foo)




On Wed, Mar 10, 2010 at 4:38 PM, Gledhill, James <
james.gledhill at sipstorm.com> wrote:

>  Thanks Anthony ?
>
>
>
> I got it to work with the following:
>
>
>
> FS Version : 16852 and later 16963M
>
>
>
> "api originate sofia/10.50.48.232/+{my_tn_here}@10.50.50.50:5060
> &conference(conf_uuid-TEST_CON)<http://10.50.48.232/+%7Bmy_tn_here%7D%4010.50.50.50%3A5060&conference%28conf_uuid-TEST_CON%29>inline"
>
>
>
> If I used *api originate <dialstring> conference conf_uuid-TEST_CONF
> inline* then FS tried to make a conference of conf_uuid-TEST_CON*@inline*.. using &conference(conf_uuid-TEST_CON) inline fixed that
>
>
>
> 2010-03-10 17:03:36.463558 [NOTICE] switch_ivr.c:1447 Transfer
> sofia/internal/+{my_tn}@{my_gw} to conf_uuid-TEST_CON[conference at inline]
>
> 2010-03-10 17:03:36.465638 [DEBUG] switch_core_state_machine.c:133 No
> Dialplan, changing state to CONSUME_MEDIA
>
>
>
>
>
> ---------
>
>
>
> For others that may look at this later, my test code to get ?start-talking?
> and ?stop-talking? events during Early Media using esl lua
> (freesiwtch_src/libs/esl/lua) is
>
>
>
>
>
> #!/usr/local/bin/lua
>
> require("ESL")
>
>
>
> local con = ESL.ESLconnection("localhost", "8021", "ClueCon");
>
>
>
> con:sendRecv("event plain ALL")
>
> ob_session_string = string.format("originate
> sofia/internal/{MY_TN_NUMBER_HERE}@{MY_GATEWAY_HERE}
> &conference(conf_uuid-TEST_CON) inline")
>
>
>
> ff = con:api( ob_session_string )
>
> print("** sent api")
>
>
>
> if ff then
>
>     print(ff:getBody())
>
>     print(ff: getHeader("result"))
>
> else
>
>     print("no body")
>
> end
>
>
>
>
>
> cnt = 0
>
> while cnt <= 40 do
>
>
>
>     print("loop = " .. cnt )
>
>     cnt = cnt + 1
>
>     ev = con:recvEventTimed(1000)
>
>     print( ev and ev:serialize("plain") or "no event this time" )
>
>
>
> end
>  ------------------------------
>
> *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, March 10, 2010 2:36 PM
>
> *To:* freeswitch-dev at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-dev] error events for failed calls
> using'conferenceconf_name bgdial sofia/ip/user'
>
>
>
> Some of your findings are incorrect.
>
> The default behavior of originate is to return as soon as there is early
> media. (sip 183)
> Adding ignore_early_media=true is the opposite effect so you do not want
> that.
>
> bgapi just does the command in a separate thread so you don't want that
> either.
>
> *api originate <dialstring> conference conf_uuid-TEST_CONF inline*
> will put the call in the conference the very instant it sees a 183
>
> If you see anything other than this, make sure you are on latest trunk and
> do a console trace demonstrating otherwise.
>
> use these commands for more details.
>
> console loglevel debug
> sofia profile internal siptrace on
>
>
>  On Wed, Mar 10, 2010 at 1:29 PM, Gledhill, James <
> james.gledhill at sipstorm.com> wrote:
>
> Anthony ?
>
>
>
> The problem with *api originate <dialstring> conference
> conf_uuid-TEST_CONF inline* is that it does not return until after the
> call is answered ? so I do not have events during early media.  I did try to
> add *{ignore_early_media=true}* but it still did not return until after I
> answered the phone (using esl lua for this test)
>
>
>
> I also tried "*bgapi originate
> {ignore_early_media=true,origination_uuid=my_uuid_my_uuid_my_test_uuid}sofia/internal/+
> 18015551212 at 10.50.40.50:5060 &conference(conf_uuid-TEST_CONF) inline*?, (I
> used a different TN, but changed it for privacy) which makes the call and
> returns events before I answer the call, BUT the conf bridge does not get
> involved (called) until AFTER the origination call is answered.
>
>
>
> Neither of these allows me to get ?start-talking? and ?stop-talking? events
> during Early Media from the conf bridge because the conference is not
> involved during the early media part of the call.
>
>
>
> Brian ?
>
>
>
> I should have been more detailed in my sofia string in the subject.  I am
> aware that sofia/ip/user is not valid. Two examples sofia string I use are
> below.
>
> * If I used sofia/10.50.48.222/1002 AND that user is registered and
> answers, then all works well.  If that user is not registered, I need to get
> USER_NOT_REGISTERED
>
> * If I use sofia/internal/+18015 at 10.50.40.50 (which is not a valid E.164
> TN) then I should get DESTINATION_OUT_OF_ORDER or some other error message.
>
>
>
>
>  ------------------------------
>
> *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, March 10, 2010 12:16 PM
> *To:* freeswitch-dev at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-dev] error events for failed calls
> using'conference conf_name bgdial sofia/ip/user'
>
>
>
> consider
>
> api originate <dialstring> conference conf_uuid-TEST_CONF inline
>
> this returns success+uuid or fail
>
> On Wed, Mar 10, 2010 at 12:04 PM, Brian West <brian at freeswitch.org> wrote:
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings
>
>
>
> Please read that there is no such thing as sofia/ip/user  (unless the ip is
> an alias on the profile)
>
>
>
> /b
>
>
>
> On Mar 10, 2010, at 11:23 AM, Gledhill, James wrote:
>
>
>
>
>
> My goals are:
>
> 1)       To get ?start-talking? and ?stop-talking? events during Early
> Media (before the call is answered) as well as after the call is answered.
>
> 2)       If there are problems with the call, get events that tell me why
> the call failed.
>
>
>
> The way I am doing it accomplishes the start/stop talking (in early media)
> BUT if the call fails, I do not get events back letting me know it failed
> and why.  I DO get CHANNEL_DESTROY and CHANNEL_STATE events that IF I had
> the UUID, I could match against, but at this point in the call I do NOT have
> that UUID AND when setting the UUID via
> ?origination_uuid=my_uuid_my_uuid_my_test_uuid?, my UUID has not yet been
> set, so as far as I can tell there is no way to match those events with
> details my application has at the time.
>
>
>
> The console shows the errors (Below are two examples of calls that failed
> in I entered ?bad? sofia locations), but they do not come back as events.
>
> 2010-03-05 14:52:20.109497 [ERR] switch_ivr_originate.c:2389 Cannot create
> outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED]
>
> 2010-03-05 16:34:13.669509 [ERR] switch_ivr_originate.c:2389 Cannot create
> outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER]
>
>
>
> So ? I am wondering is there a better/different way to do it, is there a
> work-around, should I get events to tell me the dial had problems, etc.
>
>
>
> I am using esl/lua and esl/java to experiment with this.  Both get the same
> results (as I would expect).
>
>
>
> I have had this as a ~take-a-number on IRC for about a week, so I thought I
> would add more detail and open it up to this group.  I realize many of the
> same people do both, but hoped more eyes may look at it here and give
> thoughts.
>
>
>
>
>
> ===== From Pastebin ====
>
>
>
> I am not getting 'error' events in ESL [that match a UUID I can track] when
> there is a problem with conference <conf_name> dial sofia/myip/1006, so I
> can not tell what went wrong.  Is there a work-around?
>
>
>
>  I start a call in a conference from ESL (conference conf_uuid-TEST_CONF
> bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/
> 10.50.48.232/1006)
>
>
>
>  If the 'dial' works (meaning does not get an error) then I get the events
> I want and expect.  If it errors (not able to make the call (
> [USER_NOT_REGISTERED] | [DESTINATION_OUT_OF_ORDER] | etc. ) then I do not
> get any events that tell me something went wrong.
>
>
>
>  Is there a work-around for this?
>
>
>
> =====================
>
> LUA Code  run from ./freeswitch.trunk/libs/esl/lua
>
>
>
> #!/usr/local/bin/lua
>
> require("ESL")
>
>
>
> myip = "10.50.48.239"  -- change this to your IP address
>
> myExt = 1006  -- change to one that is registered, then rerun with an
> unregistered
>
>
>
> con = ESL.ESLconnection("localhost", "8021", "ClueCon");
>
>
>
> str = string.format("conference conf_uuid-TEST_CONF bgdial
> [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/%s/%s", myip, myExt)
>
>
>
> con:sendRecv("event plain ALL")
>
>
>
>
>
> ff = con:api( str )
>
>
>
> print("** sent " .. str )
>
>
>
> print( (ff and ff:getBody() or "no body" ))
>
>
>
> cnt = 0
>
> while cnt <= 30 do
>
>
>
>     print("loop = " .. cnt )
>
>     cnt = cnt + 1
>
>     ev = con:recvEventTimed(1000)
>
>     print( ev and ev:serialize("plain") or "no event this time" )
>
>
>
> end
>
>
>
> ==============
>
>
>
> The printout is :
>
>
>
> ** sent conference conf_uuid-TEST_CONF bgdial
> [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.239/1006
>
> OK Job-UUID: a5d57b34-9205-483c-ae64-60601d8178d7
>
>
>
> loop = 0
>
> Event-Name: CHANNEL_DESTROY
>
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
>
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
>
> FreeSWITCH-IPv4: 10.50.48.232
>
> FreeSWITCH-IPv6: %3A%3A1
>
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
>
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
>
> Event-Date-Timestamp: 1267826812571550
>
> Event-Calling-File: switch_core_session.c
>
> Event-Calling-Function: switch_core_session_perform_destroy
>
> Event-Calling-Line-Number: 1051
>
> Channel-State: CS_NEW
>
> Channel-State-Number: 0
>
> Channel-Name: N/A
>
> Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66
>
> Call-Direction: outbound
>
> Presence-Call-Direction: outbound
>
> Answer-State: ringing
>
>
>
>
>
> loop = 1
>
> Event-Name: CHANNEL_STATE
>
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
>
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
>
> FreeSWITCH-IPv4: 10.50.48.232
>
> FreeSWITCH-IPv6: %3A%3A1
>
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
>
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
>
> Event-Date-Timestamp: 1267826812571550
>
> Event-Calling-File: switch_channel.c
>
> Event-Calling-Function: switch_channel_perform_set_running_state
>
> Event-Calling-Line-Number: 1301
>
> Channel-State: CS_DESTROY
>
> Channel-State-Number: 12
>
> Channel-Name: _undef_
>
> Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66
>
> Call-Direction: outbound
>
> Presence-Call-Direction: outbound
>
> Answer-State: ringing
>
>
>
>
>
> loop = 2
>
> Event-Name: API
>
> Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1
>
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
>
> FreeSWITCH-IPv4: 10.50.48.232
>
> FreeSWITCH-IPv6: %3A%3A1
>
> Event-Date-Local: 2010-03-05%2017%3A06%3A52
>
> Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT
>
> Event-Date-Timestamp: 1267826812571550
>
> Event-Calling-File: switch_loadable_module.c
>
> Event-Calling-Function: switch_api_execute
>
> Event-Calling-Line-Number: 1581
>
> API-Command: conference
>
> API-Command-Argument:
> conf_uuid-TEST_CONF%20bgdial%20%5Borigination_uuid%3Dmy_uuid_my_uuid_my_test_uuid%5Dsofia/
> 10.50.48.239/1006%20
>
>
>
>
>
> loop = 3
>
> no event this time
>
> loop = 4
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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>
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+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
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+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
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From Suneel.Papineni at mettoni.com  Thu Mar 11 09:56:03 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Thu, 11 Mar 2010 17:56:03 -0000
Subject: [Freeswitch-dev] From header with 0000000000
Message-ID: <3181A30B8C35AB4AA8577B78DDF461380696BA90@nickel.mettonigroup.com>

Hi,

 

This might be a basic question, but could you please let me know why I
am getting "0000000000" as Caller-Caller-ID-Number (This is appearing in
>From header as well) and with out From username when I am issuing
following command to originate a call.

 

api originate sofia/softphone/1004 at 172.22.252.124 9999

 

This could be a setting I need to make before making call. 

 

Thanks & Regards

Suneel 


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

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*************************************************************************

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From m.sobkow at marketelsystems.com  Thu Mar 11 10:12:43 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 11 Mar 2010 12:12:43 -0600
Subject: [Freeswitch-dev] Problem with call recording
In-Reply-To: <8D801943-7D35-4A54-8B26-791AE3333605@freeswitch.org>
References: <4B96737F.9030305@marketelsystems.com>
	<8D801943-7D35-4A54-8B26-791AE3333605@freeswitch.org>
Message-ID: <4B99329B.70807@marketelsystems.com>

Sorry, I was just hoping for confirmation that it's a bug before I filed 
a report.  I'll take care of that some time today. :)

Brian West wrote:
> Mark,
> 	Bug reports belong on Jira.
>
> 	http://jira.freeswitch.org
>
>
> /b
>
> On Mar 9, 2010, at 10:12 AM, Mark Sobkow wrote:
>
>   
>> I'm having a problem with call recording.  I tell Freeswitch to record a 
>> call to a file, and it responds ok, and the file shows up on the file 
>> system.  I later tell Freeswitch to stop recording, and it returns ok, 
>> but the file keeps growing as it doesn't actually stop recording.
>>
>> As a result, any subsequent attempts to start recording the session to a 
>> different file name fails.
>>
>> In my case, an operator is logged in to the phone system and receiving 
>> calls from customers or from automated callbacks (if the customer didn't 
>> want to wait.)  So I don't need to record the entire operator session, 
>> just _segments_ of it.
>>     
>
>
> _______________________________________________
> 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 Mar 11 10:13:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Mar 2010 12:13:24 -0600
Subject: [Freeswitch-dev] From header with 0000000000
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380696BA90@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF461380696BA90@nickel.mettonigroup.com>
Message-ID: <191c3a031003111013s123f10f6qcde6abb47026ee81@mail.gmail.com>

api originate {origination_caller_id_number=12121231234}sofia/softphone/
1004 at 172.22.252.124 9999




On Thu, Mar 11, 2010 at 11:56 AM, Suneel Papineni <
Suneel.Papineni at mettoni.com> wrote:

>  Hi,
>
>
>
> This might be a basic question, but could you please let me know why I am
> getting ?0000000000? as Caller-Caller-ID-Number (This is appearing in From
> header as well) and with out From username when I am issuing following
> command to originate a call.
>
>
>
> api originate sofia/softphone/1004 at 172.22.252.124 9999
>
>
>
> This could be a setting I need to make before making call.
>
>
>
> Thanks & Regards
>
> Suneel
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From james.gledhill at sipstorm.com  Thu Mar 11 10:28:35 2010
From: james.gledhill at sipstorm.com (Gledhill, James)
Date: Thu, 11 Mar 2010 13:28:35 -0500
Subject: [Freeswitch-dev] From header with 0000000000
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380696BA90@nickel.mettonigroup.com>
Message-ID: <A671FCA12875564E8CC11878981732BB0E17E5CB@mse2be2.mse2.exchange.ms>

That is the default setting.  

 

There are many ways to change it.  You may want to look at some of the
examples in http://wiki.freeswitch.org/wiki/Mod_commands#originate and
channel variable like those
http://wiki.freeswitch.org/wiki/Channel_Variables#Originate_related_vari
ables  

 

An example would be 

 

api originate {origination_caller_id_number=9185551212,
origination_caller_id_name=Jimmy}sofia/softphone/1004 at 172.22.252.124
9999

 

 

________________________________

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Suneel
Papineni
Sent: Thursday, March 11, 2010 11:56 AM
To: freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] From header with 0000000000

 

Hi,

 

This might be a basic question, but could you please let me know why I
am getting "0000000000" as Caller-Caller-ID-Number (This is appearing in
>From header as well) and with out From username when I am issuing
following command to originate a call.

 

api originate sofia/softphone/1004 at 172.22.252.124 9999

 

This could be a setting I need to make before making call. 

 

Thanks & Regards

Suneel 

************************************************************************
*
Please consider the environment before printing this e-mail
************************************************************************
*
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com
 
Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
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*
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From james.gledhill at sipstorm.com  Thu Mar 11 12:17:54 2010
From: james.gledhill at sipstorm.com (Gledhill, James)
Date: Thu, 11 Mar 2010 15:17:54 -0500
Subject: [Freeswitch-dev] Getting CHANNEL_ANSWER Event before call is
	answered with 'api originate <dialstring>
	conference:conf_uuid-TEST_CONF inline'
In-Reply-To: <191c3a031003111013s123f10f6qcde6abb47026ee81@mail.gmail.com>
Message-ID: <A671FCA12875564E8CC11878981732BB0E17E6BC@mse2be2.mse2.exchange.ms>

Testing with version 16963M

 

Problem:

 

When using

 * api originate sofia/internal/+18014712100 at 172.30.110.225:5060
conference:conf_uuid-TEST_CONF inline   or

 * api originate sofia/internal/+18014712100 at 172.30.110.225:5060
&conference(conf_uuid-TEST_CONF)

 


I am getting the CHANNEL_ANSWER event before the channel is actually
answered.  I have included console logs (with SIP messages) as well as
the EVENT from ESL in the enclosed log files
[Early_CHANNEL_ANSWER_EVENTS.txt & Early_CHANNEL_ANSWER_CLI_LOG.txt]

 

Is there a work-around or something else I need to do/set, or should I
log this in Jira?

 

To give an overview:

 

2010-03-11 14:30:44.785190 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/+18014712100 at 172.30.110.225:5060
[8a0782ef-fb06-4fa7-9441-ddcc65f08b4a]

2010-03-11 14:30:44.785190 [DEBUG] mod_sofia.c:3375
(sofia/internal/+18014712100 at 172.30.110.225:5060) State Change CS_NEW ->
CS_INIT

2010-03-11 14:30:44.785190 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/+18014712100 at 172.30.110.225:5060 [BREAK]

2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/+18014712100 at 172.30.110.225:5060) Running State Change
CS_INIT

2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/+18014712100 at 172.30.110.225:5060) State INIT

2010-03-11 14:30:44.785190 [DEBUG] mod_sofia.c:83
sofia/internal/+18014712100 at 172.30.110.225:5060 SOFIA INIT

2010-03-11 14:30:44.785190 [DEBUG] mod_sofia.c:117
(sofia/internal/+18014712100 at 172.30.110.225:5060) State Change CS_INIT
-> CS_ROUTING

2010-03-11 14:30:44.785190 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/+18014712100 at 172.30.110.225:5060 [BREAK]

2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/+18014712100 at 172.30.110.225:5060) State INIT going to
sleep

2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/+18014712100 at 172.30.110.225:5060) Running State Change
CS_ROUTING

2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/+18014712100 at 172.30.110.225:5060) State ROUTING

2010-03-11 14:30:44.785190 [DEBUG] mod_sofia.c:140
sofia/internal/+18014712100 at 172.30.110.225:5060 SOFIA ROUTING

2010-03-11 14:30:44.785190 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/+18014712100 at 172.30.110.225:5060) State Change
CS_ROUTING -> CS_CONSUME_MEDIA

2010-03-11 14:30:44.785190 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/+18014712100 at 172.30.110.225:5060 [BREAK]

2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/+18014712100 at 172.30.110.225:5060) State ROUTING going to
sleep

2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/+18014712100 at 172.30.110.225:5060) Running State Change
CS_CONSUME_MEDIA

2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/+18014712100 at 172.30.110.225:5060) State CONSUME_MEDIA

2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/+18014712100 at 172.30.110.225:5060) State CONSUME_MEDIA
going to sleep

send 1140 bytes to udp/[172.30.110.225]:5060 at 19:30:44.787327:

 
------------------------------------------------------------------------

   INVITE sip:+18014712100 at 172.30.110.225:5060 SIP/2.0

   <<< Deleted here for shorter version - full log in enclosed file>>>

 
------------------------------------------------------------------------

2010-03-11 14:30:44.787361 [DEBUG] sofia.c:4152 Channel
sofia/internal/+18014712100 at 172.30.110.225:5060 entering state
[calling][0]

recv 739 bytes from udp/[172.30.110.225]:5060 at 19:30:44.817367:

 
------------------------------------------------------------------------

   SIP/2.0 100 Trying

   <<< Deleted here for shorter version - full log in enclosed file>>>

 
------------------------------------------------------------------------

recv 732 bytes from udp/[172.30.110.225]:5060 at 19:30:45.667133:

 
------------------------------------------------------------------------

   SIP/2.0 180 Ringing  [NOTE: This is a 180 WITH SDP - causing Early
Media - Yes this is a Sonus!]]

   <<< Deleted here for shorter version - full log in enclosed file>>>

 
------------------------------------------------------------------------

2010-03-11 14:30:45.666976 [INFO] sofia.c:662 Update Callee ID to
"+18014712100" <+18014712100>

2010-03-11 14:30:45.724981 [DEBUG] sofia.c:4152 Channel
sofia/internal/+18014712100 at 172.30.110.225:5060 entering state
[proceeding][180]

2010-03-11 14:30:45.724981 [DEBUG] sofia.c:4163 Remote SDP:

v=0

   <<< Deleted here for shorter version - full log in enclosed file>>>

2010-03-11 14:30:45.724981 [DEBUG] sofia_glue.c:3570 Audio Codec Compare
[PCMU:0:8000:20]/[PCMU:0:8000:20]

2010-03-11 14:30:45.724981 [DEBUG] sofia_glue.c:2354 Set Codec
sofia/internal/+18014712100 at 172.30.110.225:5060 PCMU/8000 20 ms 160
samples

2010-03-11 14:30:45.727007 [DEBUG] sofia_glue.c:3511 Set 2833 dtmf
payload to 101

2010-03-11 14:30:45.727007 [DEBUG] sofia_glue.c:2594 AUDIO RTP
[sofia/internal/+18014712100 at 172.30.110.225:5060] 10.50.48.232 port
32646 -> 65.59.218.161 port 59838 codec: 0 ms: 20

2010-03-11 14:30:45.727007 [DEBUG] switch_rtp.c:1181 Starting timer
[soft] 160 bytes per 20ms

2010-03-11 14:30:45.729007 [NOTICE] sofia_glue.c:3149 Pre-Answer
sofia/internal/+18014712100 at 172.30.110.225:5060!

2010-03-11 14:30:45.730985 [DEBUG] switch_ivr_originate.c:3107 Originate
Resulted in Success: [sofia/internal/+18014712100 at 172.30.110.225:5060]

2010-03-11 14:30:45.730985 [DEBUG] switch_ivr.c:1441
(sofia/internal/+18014712100 at 172.30.110.225:5060) State Change
CS_CONSUME_MEDIA -> CS_ROUTING

   <<< Deleted here for shorter version - full log in enclosed file>>>

2010-03-11 14:30:45.730985 [DEBUG] mod_sofia.c:226
sofia/internal/+18014712100 at 172.30.110.225:5060 SOFIA EXECUTE

2010-03-11 14:30:45.730985 [DEBUG] switch_core_state_machine.c:157
sofia/internal/+18014712100 at 172.30.110.225:5060 Standard EXECUTE

EXECUTE sofia/internal/+18014712100 at 172.30.110.225:5060
conference(conf_uuid-TEST_CON)

2010-03-11 14:30:45.730985 [DEBUG] mod_conference.c:5671 Installing
default caller control action 'mute' bound to '0'.

2010-03-11 14:30:45.730985 [DEBUG] switch_ivr.c:1616 max len 1

2010-03-11 14:30:45.730985 [DEBUG] switch_ivr.c:1620 min len 1

   <<< Deleted here for shorter version - full log in enclosed file>>>

2010-03-11 14:30:45.733291 [DEBUG] switch_core_codec.c:122
sofia/internal/+18014712100 at 172.30.110.225:5060 Push codec L16:10

2010-03-11 14:30:45.733291 [DEBUG] switch_core_session.c:638 Send signal
sofia/internal/+18014712100 at 172.30.110.225:5060 [BREAK]

2010-03-11 14:30:45.733291 [DEBUG] mod_conference.c:2202 Setup timer
soft success interval: 20  samples: 160

2010-03-11 14:30:45.733291 [NOTICE] mod_conference.c:2213 Channel
[sofia/internal/+18014712100 at 172.30.110.225:5060] has been answered

  NOTE: The above line is where CHANNEL_ANSWER event gets generated, but
the call is not answered until about 10 seconds later (in this case).
See below.

2010-03-11 14:30:45.733291 [DEBUG] mod_conference.c:2316 Outbound
conference channel answered, setting CFLAG_ANSWERED

2010-03-11 14:30:45.735363 [DEBUG] mod_conference.c:994 Setup timer
success interval: 20  samples: 160

2010-03-11 14:30:46.232841 [DEBUG] switch_rtp.c:2055 Correct ip/port
confirmed.

2010-03-11 14:30:48.494337 [DEBUG] mod_local_stream.c:421 Opening Stream
[moh/8000] 8000hz

   NOTE: The 200 OK came ~10 seconds after the above "Channel
[sofia/internal/+18014712100 at 172.30.110.225:5060] has been answered"
message

recv 830 bytes from udp/[172.30.110.225]:5060 at 19:30:55.520491:

 
------------------------------------------------------------------------

   SIP/2.0 200 OK

   v:SIP/2.0/UDP
10.50.48.232;branch=z9hG4bKXQaS3DB27ypra;received=10.50.48.232;rport=506
0

   f:""<sip:0000000000 at 10.50.48.232>;tag=Bgv1HtFKD6UUK

 
t:<sip:+18014712100 at 172.30.110.225:5060>;tag=887148b37ae7121417289eefaa3
40e22

   i:6db7a397-a7e7-122d-84b6-001279948854

   <<< Deleted here for shorter version - full log in enclosed file>>>

 

 

 

>From the enclosed Event logs [Early_CHANNEL_ANSWER_EVENTS.txt]

 

loop = 17

Event-Name: CHANNEL_ANSWER

Core-UUID: 3920497f-8c61-401e-85b2-8693bcf0a428

FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com

FreeSWITCH-IPv4: 10.50.48.232

FreeSWITCH-IPv6: %3A%3A1

Event-Date-Local: 2010-03-11 14:30:45  

  [NOTE: 14:30:45] - time of the 2010-03-11 14:30:45.733291 [NOTICE]
mod_conference.c:2213 Channel
[sofia/internal/+18014712100 at 172.30.110.225:5060] has been answered

Event-Date-GMT: Thu,%2011%20Mar%202010%2019%3A30%3A45%20GMT

Event-Date-Timestamp: 1268335845733291

Event-Calling-File: switch_channel.c

Event-Calling-Function: switch_channel_perform_mark_answered

Event-Calling-Line-Number: 2283

Channel-State: CS_EXECUTE

Channel-State-Number: 4

Channel-Name: sofia/internal/%2B18014712100%40172.30.110.225%3A5060

Unique-ID: 8a0782ef-fb06-4fa7-9441-ddcc65f08b4a

Call-Direction: outbound

Presence-Call-Direction: outbound

Answer-State: answered

Channel-Read-Codec-Name: L16

Channel-Read-Codec-Rate: 8000

Channel-Write-Codec-Name: PCMU

Channel-Write-Codec-Rate: 8000

Caller-Dialplan: inline

Caller-Caller-ID-Number: 0000000000

Caller-Network-Addr: 172.30.110.225

Caller-Destination-Number: %2B18014712100

Caller-Unique-ID: 8a0782ef-fb06-4fa7-9441-ddcc65f08b4a

Caller-Source: src/switch_ivr_originate.c

Caller-Context: default

Caller-Channel-Name:
sofia/internal/%2B18014712100%40172.30.110.225%3A5060

Caller-Profile-Index: 2

Caller-Profile-Created-Time: 1268335845730985

Caller-Channel-Created-Time: 1268335844785190

Caller-Channel-Answered-Time: 1268335845733291

Caller-Channel-Progress-Time: 0

Caller-Channel-Progress-Media-Time: 1268335845729007

Caller-Channel-Hangup-Time: 0

Caller-Channel-Transfer-Time: 0

Caller-Screen-Bit: true

Caller-Privacy-Hide-Name: false

Caller-Privacy-Hide-Number: false

variable_is_outbound: true

variable_uuid: 8a0782ef-fb06-4fa7-9441-ddcc65f08b4a

variable_sip_profile_name: 10.50.48.232

variable_channel_name:
sofia/internal/%2B18014712100%40172.30.110.225%3A5060

variable_sip_destination_url:
sip%3A%2B18014712100%40172.30.110.225%3A5060

variable_originate_early_media: true

variable_sip_local_sdp_str:
v%3D0%0Ao%3DFreeSWITCH%201268303198%201268303199%20IN%20IP4%2010.50.48.2
32%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%2010.50.48.232%0At%3D0%200%0Am%3Daudi
o%2032646%20RTP/AVP%200%20101%2013%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drt
pmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Drtpmap%3
A13%20CN/8000%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A

variable_sip_outgoing_contact_uri:
%3Csip%3Amod_sofia%4010.50.48.232%3A5060%3E

variable_sip_req_uri: %2B18014712100%40172.30.110.225%3A5060

variable_sofia_profile_name: internal

variable_sip_reply_host: 172.30.110.225

variable_sip_reply_port: 5060

variable_sip_network_ip: 172.30.110.225

variable_sip_network_port: 5060

variable_sip_full_via:
SIP/2.0/UDP%2010.50.48.232%3Bbranch%3Dz9hG4bKXQaS3DB27ypra%3Breceived%3D
10.50.48.232%3Brport%3D5060

variable_sip_full_from:
%22%22%20%3Csip%3A0000000000%4010.50.48.232%3E%3Btag%3DBgv1HtFKD6UUK

variable_sip_full_to:
%3Csip%3A%2B18014712100%40172.30.110.225%3A5060%3E%3Btag%3D887148b37ae71
21417289eefaa340e22

variable_sip_from_user: 0000000000

variable_sip_from_uri: 0000000000%4010.50.48.232

variable_sip_from_host: 10.50.48.232

variable_sip_to_user: %2B18014712100

variable_sip_to_port: 5060

variable_sip_to_uri: %2B18014712100%40172.30.110.225%3A5060

variable_sip_to_host: 172.30.110.225

variable_sip_contact_params: transport%3Dudp

variable_sip_contact_user: 0-04370ee113c407043dff

variable_sip_contact_port: 5060

variable_sip_contact_uri: 0-04370ee113c407043dff%40172.30.110.225%3A5060

variable_sip_contact_host: 172.30.110.225

variable_sip_to_tag: 887148b37ae7121417289eefaa340e22

variable_sip_from_tag: Bgv1HtFKD6UUK

variable_sip_cseq: 128049586

variable_sip_call_id: 6db7a397-a7e7-122d-84b6-001279948854

variable_switch_r_sdp:
v%3D0%0D%0Ao%3DSonus_UAC%20536801%2053680100%20IN%20IP4%2065.59.218.161%
0D%0As%3DSIP%20Media%20Capabilities%0D%0Ac%3DIN%20IP4%2065.59.218.161%0D
%0At%3D0%200%0D%0Am%3Daudio%2059838%20RTP/AVP%200%20101%0D%0Aa%3Drtpmap%
3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfm
tp%3A101%200-15%0D%0Aa%3Dmaxptime%3A20%0D%0A

variable_sip_use_codec_name: PCMU

variable_sip_use_codec_rate: 8000

variable_sip_use_codec_ptime: 20

variable_write_codec: PCMU

variable_write_rate: 8000

variable_local_media_ip: 10.50.48.232

variable_local_media_port: 32646

variable_sip_use_pt: 0

variable_rtp_use_ssrc: 2908074324

variable_remote_media_ip: 65.59.218.161

variable_remote_media_port: 59838

variable_endpoint_disposition: EARLY%20MEDIA

variable_max_forwards: 70

variable_current_application_data: conf_uuid-TEST_CON

variable_current_application: conference

variable_conference_name: conf_uuid-TEST_CON

variable_read_codec: L16

variable_read_rate: 8000

 

 

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From anthony.minessale at gmail.com  Thu Mar 11 12:42:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Mar 2010 14:42:56 -0600
Subject: [Freeswitch-dev] Getting CHANNEL_ANSWER Event before call is
	answered with 'api originate <dialstring>
	conference:conf_uuid-TEST_CONF inline'
In-Reply-To: <A671FCA12875564E8CC11878981732BB0E17E6BC@mse2be2.mse2.exchange.ms>
References: <191c3a031003111013s123f10f6qcde6abb47026ee81@mail.gmail.com>
	<A671FCA12875564E8CC11878981732BB0E17E6BC@mse2be2.mse2.exchange.ms>
Message-ID: <191c3a031003111242x7a8fdf68x39bad13d6feb0f34@mail.gmail.com>

That was a very obscure bug becuse most people to not send unanswered calls
to something who might call answer on them.  its fixed in r16970 but if you
have any more bugs, please report them to http://jira.freeswitch.org not
here.


On Thu, Mar 11, 2010 at 2:17 PM, Gledhill, James <
james.gledhill at sipstorm.com> wrote:

>  Testing with version *16963M*
>
>
>
> *Problem:*
>
>
>
> When using
>
>  * api originate sofia/internal/+18014712100 at 172.30.110.225:5060conference:conf_uuid-TEST_CONF inline   or
>
>  * api originate sofia/internal/+18014712100 at 172.30.110.225:5060&conference(conf_uuid-TEST_CONF)
>
>
>
>
> I am getting the CHANNEL_ANSWER event before the channel is actually
> answered.  I have included console logs (with SIP messages) as well as the
> EVENT from ESL in the enclosed log files [Early_CHANNEL_ANSWER_EVENTS.txt &
> Early_CHANNEL_ANSWER_CLI_LOG.txt]
>
>
>
> *Is there a work-around or something else I need to do/set, or should I
> log this in Jira?*
>
>
>
> *To give an overview:*
>
>
>
> 2010-03-11 14:30:44.785190 [NOTICE] switch_channel.c:669 *New Channel
> sofia/internal/+18014712100 at 172.30.110.225:5060[8a0782ef-fb06-4fa7-9441-ddcc65f08b4a]
> *
>
> 2010-03-11 14:30:44.785190 [DEBUG] mod_sofia.c:3375 (sofia/internal/+
> 18014712100 at 172.30.110.225:5060) State Change CS_NEW -> CS_INIT
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/internal/+18014712100 at 172.30.110.225:5060 [BREAK]
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/+18014712100 at 172.30.110.225:5060) Running State Change
> CS_INIT
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/+18014712100 at 172.30.110.225:5060) State INIT
>
> 2010-03-11 14:30:44.785190 [DEBUG] mod_sofia.c:83 sofia/internal/+
> 18014712100 at 172.30.110.225:5060 SOFIA INIT
>
> 2010-03-11 14:30:44.785190 [DEBUG] mod_sofia.c:117 (sofia/internal/+
> 18014712100 at 172.30.110.225:5060) State Change CS_INIT -> CS_ROUTING
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/internal/+18014712100 at 172.30.110.225:5060 [BREAK]
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/+18014712100 at 172.30.110.225:5060) State INIT going to
> sleep
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/+18014712100 at 172.30.110.225:5060) Running State Change
> CS_ROUTING
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/+18014712100 at 172.30.110.225:5060) State ROUTING
>
> 2010-03-11 14:30:44.785190 [DEBUG] mod_sofia.c:140 sofia/internal/+
> 18014712100 at 172.30.110.225:5060 SOFIA ROUTING
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/+18014712100 at 172.30.110.225:5060) State Change CS_ROUTING
> -> CS_CONSUME_MEDIA
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/internal/+18014712100 at 172.30.110.225:5060 [BREAK]
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/+18014712100 at 172.30.110.225:5060) State ROUTING going to
> sleep
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/+18014712100 at 172.30.110.225:5060) Running State Change
> CS_CONSUME_MEDIA
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:360
> (sofia/internal/+18014712100 at 172.30.110.225:5060) State CONSUME_MEDIA
>
> 2010-03-11 14:30:44.785190 [DEBUG] switch_core_state_machine.c:360
> (sofia/internal/+18014712100 at 172.30.110.225:5060) State CONSUME_MEDIA
> going to sleep
>
> send 1140 bytes to udp/[172.30.110.225]:5060 at 19:30:44.787327:
>
>    ------------------------------------------------------------------------
>
>    INVITE sip:+18014712100 at 172.30.110.225:5060 SIP/2.0
>
> *   <<< Deleted here for shorter version ? full log in enclosed file>>>*
>
>    ------------------------------------------------------------------------
>
> 2010-03-11 14:30:44.787361 [DEBUG] sofia.c:4152 Channel sofia/internal/+
> 18014712100 at 172.30.110.225:5060 entering state [calling][0]
>
> recv 739 bytes from udp/[172.30.110.225]:5060 at 19:30:44.817367:
>
>    ------------------------------------------------------------------------
>
>    SIP/2.0 100 Trying
>
> *   <<< Deleted here for shorter version ? full log in enclosed file>>>*
>
>    ------------------------------------------------------------------------
>
> recv 732 bytes from udp/[172.30.110.225]:5060 at 19:30:45.667133:
>
>    ------------------------------------------------------------------------
>
>    SIP/2.0 180 Ringing  *[NOTE: This is a **180 WITH SDP** ? causing Early
> Media ? Yes this is a Sonus!]]*
>
> *   <<< Deleted here for shorter version ? full log in enclosed file>>>*
>
>    ------------------------------------------------------------------------
>
> 2010-03-11 14:30:45.666976 [INFO] sofia.c:662 Update Callee ID to
> "+18014712100" <+18014712100>
>
> 2010-03-11 14:30:45.724981 [DEBUG] sofia.c:4152 Channel sofia/internal/+
> 18014712100 at 172.30.110.225:5060 entering state [proceeding][180]
>
> 2010-03-11 14:30:45.724981 [DEBUG] sofia.c:4163 Remote SDP:
>
> v=0
>
> *   <<< Deleted here for shorter version ? full log in enclosed file>>>*
>
> 2010-03-11 14:30:45.724981 [DEBUG] sofia_glue.c:3570 Audio Codec Compare
> [PCMU:0:8000:20]/[PCMU:0:8000:20]
>
> 2010-03-11 14:30:45.724981 [DEBUG] sofia_glue.c:2354 Set Codec
> sofia/internal/+18014712100 at 172.30.110.225:5060 PCMU/8000 20 ms 160
> samples
>
> 2010-03-11 14:30:45.727007 [DEBUG] sofia_glue.c:3511 Set 2833 dtmf payload
> to 101
>
> 2010-03-11 14:30:45.727007 [DEBUG] sofia_glue.c:2594 AUDIO RTP
> [sofia/internal/+18014712100 at 172.30.110.225:5060] 10.50.48.232 port 32646
> -> 65.59.218.161 port 59838 codec: 0 ms: 20
>
> 2010-03-11 14:30:45.727007 [DEBUG] switch_rtp.c:1181 Starting timer [soft]
> 160 bytes per 20ms
>
> 2010-03-11 14:30:45.729007 [NOTICE] sofia_glue.c:3149 Pre-Answer
> sofia/internal/+18014712100 at 172.30.110.225:5060!
>
> 2010-03-11 14:30:45.730985 [DEBUG] switch_ivr_originate.c:3107 Originate
> Resulted in Success: [sofia/internal/+18014712100 at 172.30.110.225:5060]
>
> 2010-03-11 14:30:45.730985 [DEBUG] switch_ivr.c:1441 (sofia/internal/+
> 18014712100 at 172.30.110.225:5060) State Change CS_CONSUME_MEDIA ->
> CS_ROUTING
>
> *   <<< Deleted here for shorter version ? full log in enclosed file>>>*
>
> 2010-03-11 14:30:45.730985 [DEBUG] mod_sofia.c:226 sofia/internal/+
> 18014712100 at 172.30.110.225:5060 SOFIA EXECUTE
>
> 2010-03-11 14:30:45.730985 [DEBUG] switch_core_state_machine.c:157
> sofia/internal/+18014712100 at 172.30.110.225:5060 Standard EXECUTE
>
> *EXECUTE sofia/internal/+18014712100 at 172.30.110.225:5060conference(conf_uuid-TEST_CON)
> *
>
> 2010-03-11 14:30:45.730985 [DEBUG] mod_conference.c:5671 Installing default
> caller control action 'mute' bound to '0'.
>
> 2010-03-11 14:30:45.730985 [DEBUG] switch_ivr.c:1616 max len 1
>
> 2010-03-11 14:30:45.730985 [DEBUG] switch_ivr.c:1620 min len 1
>
> *   <<< Deleted here for shorter version ? full log in enclosed file>>>*
>
> 2010-03-11 14:30:45.733291 [DEBUG] switch_core_codec.c:122 sofia/internal/+
> 18014712100 at 172.30.110.225:5060 Push codec L16:10
>
> 2010-03-11 14:30:45.733291 [DEBUG] switch_core_session.c:638 Send signal
> sofia/internal/+18014712100 at 172.30.110.225:5060 [BREAK]
>
> 2010-03-11 14:30:45.733291 [DEBUG] mod_conference.c:2202 Setup timer soft
> success interval: 20  samples: 160
>
> *2010-03-11 14:30:45.733291 [NOTICE] mod_conference.c:2213 Channel
> [sofia/internal/+18014712100 at 172.30.110.225:5060] has been answered*
>
> *  NOTE: The above line is where CHANNEL_ANSWER event gets generated, but
> the call is not answered until about 10 seconds later (in this case).  See
> below.*
>
> 2010-03-11 14:30:45.733291 [DEBUG] mod_conference.c:2316 Outbound
> conference channel answered, setting CFLAG_ANSWERED
>
> 2010-03-11 14:30:45.735363 [DEBUG] mod_conference.c:994 Setup timer success
> interval: 20  samples: 160
>
> 2010-03-11 14:30:46.232841 [DEBUG] switch_rtp.c:2055 Correct ip/port
> confirmed.
>
> 2010-03-11 14:30:48.494337 [DEBUG] mod_local_stream.c:421 Opening Stream
> [moh/8000] 8000hz
>
> *   NOTE: The 200 OK came ~10 seconds after the above ?Channel
> [sofia/internal/+18014712100 at 172.30.110.225:5060] has been answered?
> message*
>
> recv 830 bytes from udp/[172.30.110.225]:5060 at 19:30:55.520491:
>
>    ------------------------------------------------------------------------
>
> *   SIP/2.0 200 OK*
>
>    v:SIP/2.0/UDP
> 10.50.48.232;branch=z9hG4bKXQaS3DB27ypra;received=10.50.48.232;rport=5060
>
>    f:""<sip:0000000000 at 10.50.48.232 <sip%3A0000000000 at 10.50.48.232>
> >;tag=Bgv1HtFKD6UUK
>
>    t:<sip:+18014712100 at 172.30.110.225:5060
> >;tag=887148b37ae7121417289eefaa340e22
>
>    i:6db7a397-a7e7-122d-84b6-001279948854
>
> *   <<< Deleted here for shorter version ? full log in enclosed file>>>*
>
>
>
>
>
>
>
> *From the enclosed Event logs [Early_CHANNEL_ANSWER_EVENTS.txt]*
>
>
>
> loop = 17
>
> Event-Name: *CHANNEL_ANSWER*
>
> Core-UUID: 3920497f-8c61-401e-85b2-8693bcf0a428
>
> FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com
>
> FreeSWITCH-IPv4: 10.50.48.232
>
> FreeSWITCH-IPv6: %3A%3A1
>
> Event-Date-Local: *2010-03-11* *14*:*30*:*45*
>
>   *[NOTE: 14:30:45] ? time of the 2010-03-11 14:30:45.733291 [NOTICE]
> mod_conference.c:2213 Channel [sofia/internal/+
> 18014712100 at 172.30.110.225:5060] has been answered*
>
> Event-Date-GMT: Thu,%2011%20Mar%202010%2019%3A30%3A45%20GMT
>
> Event-Date-Timestamp: 1268335845733291
>
> Event-Calling-File: switch_channel.c
>
> Event-Calling-Function: switch_channel_perform_mark_answered
>
> Event-Calling-Line-Number: 2283
>
> Channel-State: CS_EXECUTE
>
> Channel-State-Number: 4
>
> Channel-Name: sofia/internal/%2B18014712100%40172.30.110.225%3A5060
>
> Unique-ID: 8a0782ef-fb06-4fa7-9441-ddcc65f08b4a
>
> Call-Direction: outbound
>
> Presence-Call-Direction: outbound
>
> Answer-State: answered
>
> Channel-Read-Codec-Name: L16
>
> Channel-Read-Codec-Rate: 8000
>
> Channel-Write-Codec-Name: PCMU
>
> Channel-Write-Codec-Rate: 8000
>
> Caller-Dialplan: inline
>
> Caller-Caller-ID-Number: 0000000000
>
> Caller-Network-Addr: 172.30.110.225
>
> Caller-Destination-Number: %2B18014712100
>
> Caller-Unique-ID: 8a0782ef-fb06-4fa7-9441-ddcc65f08b4a
>
> Caller-Source: src/switch_ivr_originate.c
>
> Caller-Context: default
>
> Caller-Channel-Name: sofia/internal/%2B18014712100%40172.30.110.225%3A5060
>
> Caller-Profile-Index: 2
>
> Caller-Profile-Created-Time: 1268335845730985
>
> Caller-Channel-Created-Time: 1268335844785190
>
> Caller-Channel-Answered-Time: *1268335845733291*
>
> Caller-Channel-Progress-Time: 0
>
> Caller-Channel-Progress-Media-Time: 1268335845729007
>
> Caller-Channel-Hangup-Time: 0
>
> Caller-Channel-Transfer-Time: 0
>
> Caller-Screen-Bit: true
>
> Caller-Privacy-Hide-Name: false
>
> Caller-Privacy-Hide-Number: false
>
> variable_is_outbound: true
>
> variable_uuid: 8a0782ef-fb06-4fa7-9441-ddcc65f08b4a
>
> variable_sip_profile_name: 10.50.48.232
>
> variable_channel_name:
> sofia/internal/%2B18014712100%40172.30.110.225%3A5060
>
> variable_sip_destination_url: sip%3A%2B18014712100%40172.30.110.225%3A5060
>
> variable_originate_early_media: true
>
> variable_sip_local_sdp_str:
> v%3D0%0Ao%3DFreeSWITCH%201268303198%201268303199%20IN%20IP4%2010.50.48.232%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%2010.50.48.232%0At%3D0%200%0Am%3Daudio%2032646%20RTP/AVP%200%20101%2013%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Drtpmap%3A13%20CN/8000%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A
>
> variable_sip_outgoing_contact_uri:
> %3Csip%3Amod_sofia%4010.50.48.232%3A5060%3E
>
> variable_sip_req_uri: %2B18014712100%40172.30.110.225%3A5060
>
> variable_sofia_profile_name: internal
>
> variable_sip_reply_host: 172.30.110.225
>
> variable_sip_reply_port: 5060
>
> variable_sip_network_ip: 172.30.110.225
>
> variable_sip_network_port: 5060
>
> variable_sip_full_via:
> SIP/2.0/UDP%2010.50.48.232%3Bbranch%3Dz9hG4bKXQaS3DB27ypra%3Breceived%3D10.50.48.232%3Brport%3D5060
>
> variable_sip_full_from:
> %22%22%20%3Csip%3A0000000000%4010.50.48.232%3E%3Btag%3DBgv1HtFKD6UUK
>
> variable_sip_full_to:
> %3Csip%3A%2B18014712100%40172.30.110.225%3A5060%3E%3Btag%3D887148b37ae7121417289eefaa340e22
>
> variable_sip_from_user: 0000000000
>
> variable_sip_from_uri: 0000000000%4010.50.48.232
>
> variable_sip_from_host: 10.50.48.232
>
> variable_sip_to_user: %2B18014712100
>
> variable_sip_to_port: 5060
>
> variable_sip_to_uri: %2B18014712100%40172.30.110.225%3A5060
>
> variable_sip_to_host: 172.30.110.225
>
> variable_sip_contact_params: transport%3Dudp
>
> variable_sip_contact_user: 0-04370ee113c407043dff
>
> variable_sip_contact_port: 5060
>
> variable_sip_contact_uri: 0-04370ee113c407043dff%40172.30.110.225%3A5060
>
> variable_sip_contact_host: 172.30.110.225
>
> variable_sip_to_tag: 887148b37ae7121417289eefaa340e22
>
> variable_sip_from_tag: Bgv1HtFKD6UUK
>
> variable_sip_cseq: 128049586
>
> variable_sip_call_id: 6db7a397-a7e7-122d-84b6-001279948854
>
> variable_switch_r_sdp:
> v%3D0%0D%0Ao%3DSonus_UAC%20536801%2053680100%20IN%20IP4%2065.59.218.161%0D%0As%3DSIP%20Media%20Capabilities%0D%0Ac%3DIN%20IP4%2065.59.218.161%0D%0At%3D0%200%0D%0Am%3Daudio%2059838%20RTP/AVP%200%20101%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dmaxptime%3A20%0D%0A
>
> variable_sip_use_codec_name: PCMU
>
> variable_sip_use_codec_rate: 8000
>
> variable_sip_use_codec_ptime: 20
>
> variable_write_codec: PCMU
>
> variable_write_rate: 8000
>
> variable_local_media_ip: 10.50.48.232
>
> variable_local_media_port: 32646
>
> variable_sip_use_pt: 0
>
> variable_rtp_use_ssrc: 2908074324
>
> variable_remote_media_ip: 65.59.218.161
>
> variable_remote_media_port: 59838
>
> variable_endpoint_disposition: EARLY%20MEDIA
>
> variable_max_forwards: 70
>
> variable_current_application_data: conf_uuid-TEST_CON
>
> variable_current_application: conference
>
> variable_conference_name: conf_uuid-TEST_CON
>
> variable_read_codec: L16
>
> variable_read_rate: 8000
>
>
>
>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From Suneel.Papineni at mettoni.com  Fri Mar 12 07:37:11 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 12 Mar 2010 15:37:11 -0000
Subject: [Freeswitch-dev] From header with 0000000000
In-Reply-To: <191c3a031003111013s123f10f6qcde6abb47026ee81@mail.gmail.com>
References: <3181A30B8C35AB4AA8577B78DDF461380696BA90@nickel.mettonigroup.com>
	<191c3a031003111013s123f10f6qcde6abb47026ee81@mail.gmail.com>
Message-ID: <3181A30B8C35AB4AA8577B78DDF461380696BD8E@nickel.mettonigroup.com>

Thanks Anthony...

 

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 11 March 2010 18:13
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] From header with 0000000000

 

api originate
{origination_caller_id_number=12121231234}sofia/softphone/1004 at 172.22.25
2.124 9999

 

 

On Thu, Mar 11, 2010 at 11:56 AM, Suneel Papineni
<Suneel.Papineni at mettoni.com> wrote:

Hi,

 

This might be a basic question, but could you please let me know why I
am getting "0000000000" as Caller-Caller-ID-Number (This is appearing in
>From header as well) and with out From username when I am issuing
following command to originate a call.

 

api originate sofia/softphone/1004 at 172.22.252.124 9999

 

This could be a setting I need to make before making call. 

 

Thanks & Regards

Suneel 

************************************************************************
*
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************************************************************************
*
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com
 
Mettoni Ltd
Registered in England and Wales: 4485956
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_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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<mailto:sip%3A888 at conference.freeswitch.org> 
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googletalk:conf+888 at conference.freeswitch.org
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pstn:+19193869900


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*************************************************************************
This email and any files transmitted with it are confidential and
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are addressed. If you have received this email in error please notify
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From Suneel.Papineni at mettoni.com  Fri Mar 12 08:37:38 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 12 Mar 2010 16:37:38 -0000
Subject: [Freeswitch-dev] Attended Call Transfer
Message-ID: <3181A30B8C35AB4AA8577B78DDF461380696BDE6@nickel.mettonigroup.com>

Hi,

 

I am trying to establish an attended call transfer scenario as follows:

 

A call is made from extension1 to FSComm (Call 1) and is answered. Now
another call is made from FSComm to extension2 and is answered (Call 2).
I want to transfer the call between extension1 and extension2. For this
I tried to use "uuid_bridge" command at FSComm and is issued through
event sockets to FSComm as: 

"api uuid_bridge <A-leg Unique-id of Call1> <B-Leg Unique-id of Call
2>". 

This connected both the extensions and FSComm is out of loop for RTP.
This achieved call transfer feature, but my issue is as follows.

 

When I saw number of calls at FSComm (using "api show calls" command),
it has shown 0 calls, that is correct. Where as when I look for number
of channels (using "api show channels" command), it displayed as 2 with
details of current call between two extensions (i.e. A-leg of Call 1 and
B-leg of Call 2). It means FSComm didn't come out of loop completely and
is occupying the two channels. 

 

Could you please let me know how can I make FSComm to come out of loop
completely, so that I can accept/make another call at FSComm from/to
different extension. (I tried to kill the channel at FSComm, but this is
clearing the call between extensions completely)

 

I tried to use "uuid_deflect" command, but that is useful in Blind
transfer and not in attended transfer. Also tried with "uuid_transfer"
command but failed to achieve.

 

Could someone please let me know if there is any other method to achieve
this attended transfer scenario.

 

Thanks & Regards

Suneel

 


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

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From anthony.minessale at gmail.com  Fri Mar 12 08:52:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Mar 2010 10:52:19 -0600
Subject: [Freeswitch-dev] Attended Call Transfer
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380696BDE6@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF461380696BDE6@nickel.mettonigroup.com>
Message-ID: <191c3a031003120852j3ed667d7m4f640bdfd8c14750@mail.gmail.com>

try uuid_simplify <uuid> instead


On Fri, Mar 12, 2010 at 10:37 AM, Suneel Papineni <
Suneel.Papineni at mettoni.com> wrote:

>  Hi,
>
>
>
> I am trying to establish an attended call transfer scenario as follows:
>
>
>
> A call is made from extension1 to FSComm (Call 1) and is answered. Now
> another call is made from FSComm to extension2 and is answered (Call 2). I
> want to transfer the call between extension1 and extension2. For this I
> tried to use ?uuid_bridge? command at FSComm and is issued through event
> sockets to FSComm as:
>
> "api uuid_bridge <A-leg Unique-id of Call1> <B-Leg Unique-id of Call 2>".
>
> This connected both the extensions and FSComm is out of loop for RTP. This
> achieved call transfer feature, but my issue is as follows.
>
>
>
> When I saw number of calls at FSComm (using ?api show calls? command), it
> has shown 0 calls, that is correct. Where as when I look for number of
> channels (using ?api show channels? command), it displayed as 2 with details
> of current call between two extensions (i.e. A-leg of Call 1 and B-leg of
> Call 2). It means FSComm didn't come out of loop completely and is occupying
> the two channels.
>
>
>
> Could you please let me know how can I make FSComm to come out of loop
> completely, so that I can accept/make another call at FSComm from/to
> different extension. (I tried to kill the channel at FSComm, but this is
> clearing the call between extensions completely)
>
>
>
> I tried to use ?uuid_deflect? command, but that is useful in Blind transfer
> and not in attended transfer. Also tried with ?uuid_transfer? command but
> failed to achieve.
>
>
>
> Could someone please let me know if there is any other method to achieve
> this attended transfer scenario.
>
>
>
> Thanks & Regards
>
> Suneel
>
>
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From m.sobkow at marketelsystems.com  Sat Mar 13 17:00:10 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Sat, 13 Mar 2010 19:00:10 -0600
Subject: [Freeswitch-dev] JIRA Commented: (DP-15) uuid_record not working
In-Reply-To: <2117451631.1268525585069.JavaMail.root@jira-01>
References: <2117451631.1268525585069.JavaMail.root@jira-01>
Message-ID: <4B9C351A.1020207@marketelsystems.com>

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From brian at freeswitch.org  Sat Mar 13 17:12:05 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Mar 2010 19:12:05 -0600
Subject: [Freeswitch-dev] JIRA Commented: (DP-15) uuid_record not working
In-Reply-To: <4B9C351A.1020207@marketelsystems.com>
References: <2117451631.1268525585069.JavaMail.root@jira-01>
	<4B9C351A.1020207@marketelsystems.com>
Message-ID: <6E00DF76-2E51-479F-9F22-7969246DC31C@freeswitch.org>

Chances are you're doing something silly like trying to uuid_record a proxy_media session or a bypass media session .. I can 100% confirm that uuid_record does work.

/b

On Mar 13, 2010, at 7:00 PM, Mark Sobkow wrote:

> Anthony Minessale II (JIRA) wrote:
>> 
>>     [ http://jira.freeswitch.org/browse/DP-15?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=19666#action_19666 ] 
>> 
>> Anthony Minessale II commented on DP-15:
>> ----------------------------------------
>> 
>> I would appriciate a little more leg work than this.
>> What is this output from that you pasted here?
>>   
> It's the debug output produced by the syslog:debug statements in the code fragment I included.
>> Did you capture the exact command you are actually sending to fs?
>>   
> api( uuid_record, "OperatorUUID /var/spool/pursuit/pursuit-20100312_172020.gsm" )
>> Do you have any debug output from fs itself on debug log level?
>>   
> I run Freeswitch with full debugging enabled, and had fs_cli open at the time.  However, it didn't produce any messages that seemed relevant, such as the start recording events that I used to see with the previous SVN.
>> I am going to guess there is a lot more info you could have captured before dumping your issue off on our doorstep.
>> 
>>   
> Sorry if it seems that way.  I did spend over a day updating Freeswitch to the latest SVN trunk, was informed of the need to upgrade Wanpipe along the way, wrassled with that (mainly due to our build system, not because of Freeswitch), and then retested this morning.
> 
> I could include gobs of logfile output about the calls being established, the UUIDs associated with them, and a walkthrough about what I was doing with the phone system at the time, but that seemed to be information overload that had little to do with the core problem.
> 
> In short, the operator logs in to the system with a PIN code and is put on hold waiting for a customer to call in or for an automated callback to be processed.  They're then bridged to the customer (works), and I try to start recording the call with uuid_record.  When they're done the call (customer hangs up), the operator call is redirected to a play_and_get_digits dialplan fragment, which routes the result code to another Erlang callback that calculates the duration of the call, records the result code, and issues a uuid_record stop command before placing the operator back on hold, waiting for the next customer.
> 
> I'd be happy to rerun the tests and capture the full fs_cli log for you, if it would be of help.  However, as I mentioned, I didn't see anything in those log statements about call recordings.  With over twenty years of programming under my belt, I do understand the importance of providing relevant information to debug a problem, but I also know better than to just blindly attach entire log files full of what is essentially garbage relevant to the problem.
>>> uuid_record not working
>>> -----------------------
>>> 
>>>                 Key: DP-15
>>>                 URL: http://jira.freeswitch.org/browse/DP-15
>>>             Project: Dialplan Modules
>>>          Issue Type: Bug
>>>         Environment: Ubuntu Hardy 32-bit
>>>            Reporter: Mark Sobkow
>>>            Assignee: Anthony Minessale II
>>>            Priority: Major
>>> 
>>> When I try to record a call as follows, I get an error from Freeswitch.  With svn16569, it would do the first recording request and error out on subsequent requests for a long-lived operator session that gets bridged to multiple customers over the  course of a day.  Now it's refusing to record even the first session with svn16972.
>>> recordCall( Operator ) ->
>>> 	{ ok, DialerSpoolMsgDir } = mnesia_sofia:getSofiaConfigValue( dialerSpoolMsgDir ),
>>> 	{ ok, DialerSpoolMsgPrefix } = mnesia_sofia:getSofiaConfigValue( dialerSpoolMsgPrefix ),
>>> 	{ ok, DialerSpoolMsgType } = mnesia_sofia:getSofiaConfigValue( dialerSpoolMsgType ),
>>> 	{{ Year, Month, Day }, { Hour, Min, Sec }} = erlang:localtime(),
>>> 	Timestamp = lists:flatten(
>>> 		io_lib:format( "~4.10.0B~2.10.0B~2.10.0B_~2.10.0B~2.10.0B~2.10.0B",
>>> 			[Year, Month, Day, Hour, Min, Sec] ) ),
>>> 	FullFileName = DialerSpoolMsgDir ++ "/"
>>> 		++ DialerSpoolMsgPrefix ++ "-" ++ Timestamp ++ "." ++ DialerSpoolMsgType,
>>> 	syslog:info( "pbx_bridge:recordCall() Recording call to ~s~n", [FullFileName] ),
>>> 	RecordResult = pbx:api( uuid_record, Operator#pbx_operator_registry.operator_uuid ++ " start " ++ FullFileName ),
>>> 	syslog:debug( "pbx_bridge:recordCall() RecordResult is ~p~n", [RecordResult] ),
>>> 	Operator#pbx_operator_registry{ record_file_name=FullFileName }.
>>> 2010-03-12 17:20:20 DEBUG <0.645.0>: pbx_bridge:bridgeOperatorAndCustomer() Record all calls is true
>>> 2010-03-12 17:20:20 INFO <0.645.0>: pbx_bridge:recordCall() Recording call to /var/spool/pursuit/pursuit-20100312_172020.gsm
>>> 2010-03-12 17:20:20 DEBUG <0.645.0>: pbx:api() Sent to Pid <0.894.0>
>>> 2010-03-12 17:20:20 DEBUG <0.645.0>: pbx_bridge:recordCall() RecordResult is {error,
>>>                                                                               "-ERR Cannot record session!\n"}
>>>     
>> 
>>   
> 
> _______________________________________________
> 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
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From gmaruzz at celliax.org  Mon Mar 15 01:15:55 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 15 Mar 2010 09:15:55 +0100
Subject: [Freeswitch-dev] Skypiax has been renamed as Skypopen
Message-ID: <7b197bef1003150115la6f9540l77e0de601d1ce0f9@mail.gmail.com>

Ciao FreeSWITCHers,

is with pleasure that we're announcing the renaming of mod_skypiax
module to mod_skypopen.

The new wiki page is at
http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk

To upgrade from a previous Skypiax installation, please read:
http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Upgrading_from_Skypiax

Many many many thanks to the wonderful FreeSWITCH team that make it all happens!

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From Suneel.Papineni at mettoni.com  Mon Mar 15 10:08:43 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Mon, 15 Mar 2010 17:08:43 -0000
Subject: [Freeswitch-dev] Attended Call Transfer
In-Reply-To: <191c3a031003120852j3ed667d7m4f640bdfd8c14750@mail.gmail.com>
References: <3181A30B8C35AB4AA8577B78DDF461380696BDE6@nickel.mettonigroup.com>
	<191c3a031003120852j3ed667d7m4f640bdfd8c14750@mail.gmail.com>
Message-ID: <3181A30B8C35AB4AA8577B78DDF461380696C185@nickel.mettonigroup.com>

Hi Anthony,

 

Could you please explain me how can use uuid_simlify to achieve Attended
call transfer. I tried with many combinations, but unable to achieve
what I have explained below.

 

Or guide me to respective wiki page. Mod_Commands wiki page doesn't
explain much about this command.

 

Thanks & Regards

Suneel

 

 

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 12 March 2010 16:52
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] Attended Call Transfer

 

try uuid_simplify <uuid> instead



On Fri, Mar 12, 2010 at 10:37 AM, Suneel Papineni
<Suneel.Papineni at mettoni.com> wrote:

Hi,

 

I am trying to establish an attended call transfer scenario as follows:

 

A call is made from extension1 to FSComm (Call 1) and is answered. Now
another call is made from FSComm to extension2 and is answered (Call 2).
I want to transfer the call between extension1 and extension2. For this
I tried to use "uuid_bridge" command at FSComm and is issued through
event sockets to FSComm as: 

"api uuid_bridge <A-leg Unique-id of Call1> <B-Leg Unique-id of Call
2>". 

This connected both the extensions and FSComm is out of loop for RTP.
This achieved call transfer feature, but my issue is as follows.

 

When I saw number of calls at FSComm (using "api show calls" command),
it has shown 0 calls, that is correct. Where as when I look for number
of channels (using "api show channels" command), it displayed as 2 with
details of current call between two extensions (i.e. A-leg of Call 1 and
B-leg of Call 2). It means FSComm didn't come out of loop completely and
is occupying the two channels. 

 

Could you please let me know how can I make FSComm to come out of loop
completely, so that I can accept/make another call at FSComm from/to
different extension. (I tried to kill the channel at FSComm, but this is
clearing the call between extensions completely)

 

I tried to use "uuid_deflect" command, but that is useful in Blind
transfer and not in attended transfer. Also tried with "uuid_transfer"
command but failed to achieve.

 

Could someone please let me know if there is any other method to achieve
this attended transfer scenario.

 

Thanks & Regards

Suneel

 

************************************************************************
*
Please consider the environment before printing this e-mail
************************************************************************
*
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com
 
Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
************************************************************************
*


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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

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From anthony.minessale at gmail.com  Mon Mar 15 10:34:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 11:34:00 -0600
Subject: [Freeswitch-dev] Attended Call Transfer
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380696C185@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF461380696BDE6@nickel.mettonigroup.com>
	<191c3a031003120852j3ed667d7m4f640bdfd8c14750@mail.gmail.com>
	<3181A30B8C35AB4AA8577B78DDF461380696C185@nickel.mettonigroup.com>
Message-ID: <191c3a031003151034o6dd664e4wb1652341c1295ff1@mail.gmail.com>

If you have a bridge spanning 2 boxes

phone1 ----> |FS1|  ----> |FS2|
phone2 <---- |FS1| <----- |FS2|

you can run uuid_simplify on either uuid on FS1 to do a refer+replaces to
the 2nd FS
so the call is bridged on that box and both legs on your box1 are hungup.

phone1 -----> |FS2|
phone2 <----- |FS2|

On Mon, Mar 15, 2010 at 11:08 AM, Suneel Papineni <
Suneel.Papineni at mettoni.com> wrote:

>  Hi Anthony,
>
>
>
> Could you please explain me how can use uuid_simlify to achieve Attended
> call transfer. I tried with many combinations, but unable to achieve what I
> have explained below.
>
>
>
> Or guide me to respective wiki page. Mod_Commands wiki page doesn?t explain
> much about this command.
>
>
>
> Thanks & Regards
>
> Suneel
>
>
>
>
>
> *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* 12 March 2010 16:52
> *To:* freeswitch-dev at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-dev] Attended Call Transfer
>
>
>
> try uuid_simplify <uuid> instead
>
>  On Fri, Mar 12, 2010 at 10:37 AM, Suneel Papineni <
> Suneel.Papineni at mettoni.com> wrote:
>
> Hi,
>
>
>
> I am trying to establish an attended call transfer scenario as follows:
>
>
>
> A call is made from extension1 to FSComm (Call 1) and is answered. Now
> another call is made from FSComm to extension2 and is answered (Call 2). I
> want to transfer the call between extension1 and extension2. For this I
> tried to use ?uuid_bridge? command at FSComm and is issued through event
> sockets to FSComm as:
>
> "api uuid_bridge <A-leg Unique-id of Call1> <B-Leg Unique-id of Call 2>".
>
> This connected both the extensions and FSComm is out of loop for RTP. This
> achieved call transfer feature, but my issue is as follows.
>
>
>
> When I saw number of calls at FSComm (using ?api show calls? command), it
> has shown 0 calls, that is correct. Where as when I look for number of
> channels (using ?api show channels? command), it displayed as 2 with details
> of current call between two extensions (i.e. A-leg of Call 1 and B-leg of
> Call 2). It means FSComm didn't come out of loop completely and is occupying
> the two channels.
>
>
>
> Could you please let me know how can I make FSComm to come out of loop
> completely, so that I can accept/make another call at FSComm from/to
> different extension. (I tried to kill the channel at FSComm, but this is
> clearing the call between extensions completely)
>
>
>
> I tried to use ?uuid_deflect? command, but that is useful in Blind transfer
> and not in attended transfer. Also tried with ?uuid_transfer? command but
> failed to achieve.
>
>
>
> Could someone please let me know if there is any other method to achieve
> this attended transfer scenario.
>
>
>
> Thanks & Regards
>
> Suneel
>
>
>
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>
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>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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> IRC: irc.freenode.net #freeswitch
>
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> *************************************************************************
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> *************************************************************************
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>
> _______________________________________________
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> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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From waleedot2003 at gmail.com  Mon Mar 15 04:29:52 2010
From: waleedot2003 at gmail.com (Waleed)
Date: Mon, 15 Mar 2010 14:29:52 +0300
Subject: [Freeswitch-dev] Offline Message Module
Message-ID: <6a29f4391003150429p3c31a69aj9c7f13c06ad4b958@mail.gmail.com>

hello All
i want to build application module that let FS to send IM message to the
user when another user try to call him and he is Off-line or not registered
now , can any one help me  ? :)

thanks & regards
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From msc at freeswitch.org  Mon Mar 15 11:52:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:52:31 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conf Call Agenda
Message-ID: <87f2f3b91003151152s1acc22f4ydf45f2e6630673b3@mail.gmail.com>

Hello all,

The agenda for Wednesday's conference call is now up:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_03_17

We could use some more content. Also, we would love to have a volunteer
discuss one of the FreeSWITCH modules or some other FS feature that he/she
uses. If you have any thoughts or suggestions please email me off list.

Thanks!
-MC
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From scaram at hotmail.de  Tue Mar 16 04:47:38 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Tue, 16 Mar 2010 12:47:38 +0100
Subject: [Freeswitch-dev] hooked functions are not called while using one
	single session
Message-ID: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>


Hello,
I am writing my own freeswitch module and have implemented 2 hooks for reading and writing audio (I did not implement the endpoint interface).
My problem is that the channel_write_frame and channel_read_frame callback-functions are only called if I make an intercept with another session (switch_ivr_intercept_session), but my usecase has only one single Session for getting and putting audio data in my module. 

How can I achieve this? 
scaram

/* on channel init I answer the session and set hooks */
static switch_status_t channel_on_init(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);
    switch_channel_answer(channel);

    switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
    switch_set_flag(tech_pvt, TFLAG_IO);

    switch_core_event_hook_add_write_frame(session, &channel_write_frame);
    switch_core_event_hook_add_read_frame(session, &channel_read_frame);
}

static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
    /* is only called when I intercept 2 Sessions */
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
    /* is only called when I intercept 2 Sessions */
}


 		 	   		  
_________________________________________________________________
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From anthony.minessale at gmail.com  Tue Mar 16 06:34:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 08:34:18 -0500
Subject: [Freeswitch-dev] hooked functions are not called while using
	one single session
In-Reply-To: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>
Message-ID: <191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>

What exactly is your goal?

those hooks will probably only be called if you send the channel to do
something that
requires reading and writing audio.


On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:

>  Hello,
> I am writing my own freeswitch module and have implemented 2 hooks for
> reading and writing audio (I did not implement the endpoint interface).
> My problem is that the channel_write_frame and channel_read_frame
> callback-functions are only called if I make an intercept with another
> session (switch_ivr_intercept_session), but my usecase has only one single
> Session for getting and putting audio data in my module.
>
> How can I achieve this?
> scaram
>
> /* on channel init I answer the session and set hooks */
> static switch_status_t channel_on_init(switch_core_session_t *session)
> {
>     switch_channel_t *channel = switch_core_session_get_channel(session);
>     switch_channel_answer(channel);
>
>     switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
>     switch_set_flag(tech_pvt, TFLAG_IO);
>
>     switch_core_event_hook_add_write_frame(session, &channel_write_frame);
>     switch_core_event_hook_add_read_frame(session, &channel_read_frame);
> }
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
>
>
>
> ------------------------------
> Ein Postfach f?r Alles ?   bei Hotmail 5 E-Mail-Adressen online verwalten!<http://redirect.gimas.net/?n=M1003HM5Adressen>
>
> _______________________________________________
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> 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 <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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From scaram at hotmail.de  Tue Mar 16 06:51:08 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Tue, 16 Mar 2010 14:51:08 +0100
Subject: [Freeswitch-dev] hooked functions are not called while
	using	one single session
In-Reply-To: <191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>,
	<191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>
Message-ID: <SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>


The module I am writing should be an endpoint VOIP-interface for another program. The module should answer a call and transport incoming audio into a 3rd party application. Furthermore the 3rd party application sends audio data back to the caller. This is why I hooked channel_read_frame and channel_write_frame. 

How can I trigger the channel to do something that requires reading or writing audio? Does it make a difference to use the endpoint interface instead of the hooks? 


Date: Tue, 16 Mar 2010 08:34:18 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

What exactly is your goal?

those hooks will probably only be called if you send the channel to do something that 
requires reading and writing audio.


On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:






Hello,
I am writing my own freeswitch module and have implemented 2 hooks for reading and writing audio (I did not implement the endpoint interface).
My problem is that the channel_write_frame and channel_read_frame callback-functions are only called if I make an intercept with another session (switch_ivr_intercept_session), but my usecase has only one single Session for getting and putting audio data in my module. 


How can I achieve this? 
scaram

/* on channel init I answer the session and set hooks */
static switch_status_t channel_on_init(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);

    switch_channel_answer(channel);

    switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
    switch_set_flag(tech_pvt, TFLAG_IO);

    switch_core_event_hook_add_write_frame(session, &channel_write_frame);

    switch_core_event_hook_add_read_frame(session, &channel_read_frame);
}

static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)

{
    /* is only called when I intercept 2 Sessions */
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
    /* is only called when I intercept 2 Sessions */

}


 		 	   		  
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_______________________________________________

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

Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
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From Claudio.Cavalera at italtel.it  Tue Mar 16 08:11:18 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Tue, 16 Mar 2010 16:11:18 +0100
Subject: [Freeswitch-dev] C library and threads within openwrt
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F36420@BESONE.corp.dom>

Hello,
I'm still working with freeswitch inside openwrt build system.

What C library do you advise among these possibilities:
1) uclibc with linuxthreads-old
2) uclibc with NPTL
3) glibc (I guess this implements NPTL)

I'm asking this because I know freeswitch heavy relies on threads and
I've experienced some fun behaviour :-)

Thanks,
Claudio


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From tculjaga at gmail.com  Tue Mar 16 10:04:10 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 18:04:10 +0100
Subject: [Freeswitch-dev] stop/start RTP
Message-ID: <65d96fc81003161004j614f8c6du6c4007b09835d385@mail.gmail.com>

is there any way to stop/start RTP traffic on an active call?


e.g. some API call or event that can be triggered through event socket ?



T.
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From anthony.minessale at gmail.com  Tue Mar 16 10:15:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 12:15:03 -0500
Subject: [Freeswitch-dev] stop/start RTP
In-Reply-To: <65d96fc81003161004j614f8c6du6c4007b09835d385@mail.gmail.com>
References: <65d96fc81003161004j614f8c6du6c4007b09835d385@mail.gmail.com>
Message-ID: <191c3a031003161015r7704b01k77cdb6c36536fbf3@mail.gmail.com>

uuid_hold might


On Tue, Mar 16, 2010 at 12:04 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> is there any way to stop/start RTP traffic on an active call?
>
>
> e.g. some API call or event that can be triggered through event socket ?
>
>
>
> T.
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Tue Mar 16 11:59:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 13:59:28 -0500
Subject: [Freeswitch-dev] hooked functions are not called while using
	one single session
In-Reply-To: <SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>
	<191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>
	<SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>
Message-ID: <191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>

you probably want to use the application interface and a while loop while
reading a frame.

see most of the functions in switch_ivr_play_say


On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:

>  The module I am writing should be an endpoint VOIP-interface for another
> program. The module should answer a call and transport incoming audio into a
> 3rd party application. Furthermore the 3rd party application sends audio
> data back to the caller. This is why I hooked channel_read_frame and
> channel_write_frame.
>
> How can I trigger the channel to do something that requires reading or
> writing audio? Does it make a difference to use the endpoint interface
> instead of the hooks?
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 08:34:18 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
>
> What exactly is your goal?
>
> those hooks will probably only be called if you send the channel to do
> something that
> requires reading and writing audio.
>
>
> On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  Hello,
> I am writing my own freeswitch module and have implemented 2 hooks for
> reading and writing audio (I did not implement the endpoint interface).
> My problem is that the channel_write_frame and channel_read_frame
> callback-functions are only called if I make an intercept with another
> session (switch_ivr_intercept_session), but my usecase has only one single
> Session for getting and putting audio data in my module.
>
> How can I achieve this?
> scaram
>
> /* on channel init I answer the session and set hooks */
> static switch_status_t channel_on_init(switch_core_session_t *session)
> {
>     switch_channel_t *channel = switch_core_session_get_channel(session);
>     switch_channel_answer(channel);
>
>     switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
>     switch_set_flag(tech_pvt, TFLAG_IO);
>
>     switch_core_event_hook_add_write_frame(session, &channel_write_frame);
>     switch_core_event_hook_add_read_frame(session, &channel_read_frame);
> }
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
>
>
>
> ------------------------------
> Ein Postfach f?r Alles ?   bei Hotmail 5 E-Mail-Adressen online verwalten!<http://redirect.gimas.net/?n=M1003HM5Adressen>
>
> _______________________________________________
> 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
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From james.gledhill at sipstorm.com  Tue Mar 16 14:03:22 2010
From: james.gledhill at sipstorm.com (Gledhill, James)
Date: Tue, 16 Mar 2010 17:03:22 -0400
Subject: [Freeswitch-dev] esljni.dll
In-Reply-To: <191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>
Message-ID: <A671FCA12875564E8CC11878981732BB0E20C590@mse2be2.mse2.exchange.ms>

Does anyone have a esljni.dll already built they can send me.  I do not
have the background to build it on Windows, but want to run it from
Windows.  I have it working on Linux, but need the dll for Windows.

 

Thanks

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From brian at freeswitch.org  Tue Mar 16 14:11:06 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 16:11:06 -0500
Subject: [Freeswitch-dev] esljni.dll
In-Reply-To: <A671FCA12875564E8CC11878981732BB0E20C590@mse2be2.mse2.exchange.ms>
References: <A671FCA12875564E8CC11878981732BB0E20C590@mse2be2.mse2.exchange.ms>
Message-ID: <C0123F57-3B8D-414B-BB12-C8E5880208A7@freeswitch.org>

I suspect its in the JDK/JNI dev kit you get from sun/oracle.  Also DO NOT HIJACK threads.  If you compose an email to this list please click NEW then input the address and send it.  What you did was take an existing message and click reply, change the subject and delete the body.  Please don't do that moving forward if you can keep from it.

/b

On Mar 16, 2010, at 4:03 PM, Gledhill, James wrote:

> Does anyone have a esljni.dll already built they can send me.  I do not have the background to build it on Windows, but want to run it from Windows.  I have it working on Linux, but need the dll for Windows.
>  
> Thanks

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From tculjaga at gmail.com  Tue Mar 16 14:28:11 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 22:28:11 +0100
Subject: [Freeswitch-dev] stop/start RTP
In-Reply-To: <191c3a031003161015r7704b01k77cdb6c36536fbf3@mail.gmail.com>
References: <65d96fc81003161004j614f8c6du6c4007b09835d385@mail.gmail.com>
	<191c3a031003161015r7704b01k77cdb6c36536fbf3@mail.gmail.com>
Message-ID: <65d96fc81003161428r67c22557kbe4fb7ce44918099@mail.gmail.com>

On Tue, Mar 16, 2010 at 6:15 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> uuid_hold might
>
>
>
> Where do you store the information about established calls? Is it a core
database or cache?

lets suppose we have 2 FS servers sitting on top of a DB cluster. The left
server has a floating (public) IP. This floating IP can be moved to the
right server... according to the DB cluster, the right server should be
aware of all ongoing established sessions so, we could send re-invite/update
to both legs of every established call and perhaps use uuid_hold to make it
start sending RTP?

feasible?

T.
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From james.gledhill at sipstorm.com  Tue Mar 16 14:46:33 2010
From: james.gledhill at sipstorm.com (Gledhill, James)
Date: Tue, 16 Mar 2010 17:46:33 -0400
Subject: [Freeswitch-dev] esljni.dll
In-Reply-To: <C0123F57-3B8D-414B-BB12-C8E5880208A7@freeswitch.org>
Message-ID: <A671FCA12875564E8CC11878981732BB0E20C5DC@mse2be2.mse2.exchange.ms>

What I am looking for is someone that has compiled the FreeSwitch Event
Socket Library for Windows (fs_src/libs/esl/java -- and created the
windows dll required to use with Java).  To make JNI stuff work you need
the java classes (independent of OS) and the JNI library (OS dependant).
I can make and use the linux library, but not the Windows version.  I
was hoping someone had built it on Windows and could share.   The VC++
2008 stuff in that directory (src/libs/esl/) does not compile/link in
the library for Windows.

 

Sorry about the HIJACK.  I was not aware.  I will do differently (new vs
reply) going forward.

 

________________________________

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, March 16, 2010 4:11 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] esljni.dll

 

I suspect its in the JDK/JNI dev kit you get from sun/oracle.  Also DO
NOT HIJACK threads.  If you compose an email to this list please click
NEW then input the address and send it.  What you did was take an
existing message and click reply, change the subject and delete the
body.  Please don't do that moving forward if you can keep from it.

 

/b

 

On Mar 16, 2010, at 4:03 PM, Gledhill, James wrote:





Does anyone have a esljni.dll already built they can send me.  I do not
have the background to build it on Windows, but want to run it from
Windows.  I have it working on Linux, but need the dll for Windows.

 

Thanks

 

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From vetali100 at gmail.com  Tue Mar 16 10:19:03 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 19:19:03 +0200
Subject: [Freeswitch-dev] stop/start RTP
In-Reply-To: <65d96fc81003161004j614f8c6du6c4007b09835d385@mail.gmail.com>
References: <65d96fc81003161004j614f8c6du6c4007b09835d385@mail.gmail.com>
Message-ID: <a858e7691003161019s3b6029e4p13c7892fd413f777@mail.gmail.com>

can't this be used?

uuid_media off <uuid>


http://wiki.freeswitch.org/wiki/Bypass_Media#How_to_disable.2Fenable_it_on_the_fly.3F


2010/3/16 Tihomir Culjaga <tculjaga at gmail.com>

> is there any way to stop/start RTP traffic on an active call?
>
>
> e.g. some API call or event that can be triggered through event socket ?
>
>
>
> T.
>
> _______________________________________________
> 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 brian at freeswitch.org  Wed Mar 17 06:10:04 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 08:10:04 -0500
Subject: [Freeswitch-dev] stop/start RTP
In-Reply-To: <a858e7691003161019s3b6029e4p13c7892fd413f777@mail.gmail.com>
References: <65d96fc81003161004j614f8c6du6c4007b09835d385@mail.gmail.com>
	<a858e7691003161019s3b6029e4p13c7892fd413f777@mail.gmail.com>
Message-ID: <59A42677-D168-455C-BBB6-64E22A66CBE8@freeswitch.org>

No thats for bypass media on and off on demand.

/b

On Mar 16, 2010, at 12:19 PM, Vitalii Colosov wrote:

> can't this be used?
> uuid_media off <uuid>
> 
> http://wiki.freeswitch.org/wiki/Bypass_Media#How_to_disable.2Fenable_it_on_the_fly.3F

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From m.sobkow at marketelsystems.com  Wed Mar 17 11:29:25 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Wed, 17 Mar 2010 12:29:25 -0600
Subject: [Freeswitch-dev] Woohoo!  We have call recording working!
Message-ID: <4BA11F85.8090803@marketelsystems.com>

As per Andrew Thompson's suggestion, I changed the processing of the 
call recording to start recording as part of processing a CHANNEL_BRIDGE 
event.  It turns out that (at least with Erlang) the call is not 
actually bridged when the api command to bridge a call returns ok; the 
ok only means the bridge command was successfully queued for processing, 
not that it's completed.

I guess I wouldn't call this a bug with call recording, but it's 
definitely a "gotcha".

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From vipkilla at gmail.com  Wed Mar 17 14:23:02 2010
From: vipkilla at gmail.com (vip killa)
Date: Wed, 17 Mar 2010 16:23:02 -0500
Subject: [Freeswitch-dev] FSComm and CELT incompatible destination
Message-ID: <957f61371003171423r11ccc9f3pf5a96a9d0e2c7f89@mail.gmail.com>

when i try to connect a call from FSComm to Freeswitch using the CELT codec
i get incompatible destination.
Is this because Freeswitch uses CELT 0.7.1 and FSComm uses a previous
version? if this is the reason can someone update FSComm to CELT 0.7.1?
Thanks.
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From william.suffill at gmail.com  Wed Mar 17 14:40:04 2010
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 17 Mar 2010 17:40:04 -0400
Subject: [Freeswitch-dev] FSComm and CELT incompatible destination
In-Reply-To: <957f61371003171423r11ccc9f3pf5a96a9d0e2c7f89@mail.gmail.com>
References: <957f61371003171423r11ccc9f3pf5a96a9d0e2c7f89@mail.gmail.com>
Message-ID: <6b65470d1003171440i3452a6f0gf1b80c79dd3a0aa4@mail.gmail.com>

FSComm is actually an application that runs an embedded copy of the
FreeSwitch core.

Best to run fs_cli on both the box running FSComm and the freeswitch server
and capture debug logs so we can more properly assist in diagnosing your
problem.

-- W
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From brian at freeswitch.org  Wed Mar 17 14:48:31 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 16:48:31 -0500
Subject: [Freeswitch-dev] FSComm and CELT incompatible destination
In-Reply-To: <6b65470d1003171440i3452a6f0gf1b80c79dd3a0aa4@mail.gmail.com>
References: <957f61371003171423r11ccc9f3pf5a96a9d0e2c7f89@mail.gmail.com>
	<6b65470d1003171440i3452a6f0gf1b80c79dd3a0aa4@mail.gmail.com>
Message-ID: <2866EADB-C021-4511-B8A1-93BADA436439@freeswitch.org>

Chances are its the 10ms ptime ... see if your first codec list on the fs side is 20ms all others are removed from the list. 

/b

On Mar 17, 2010, at 4:40 PM, William Suffill wrote:

> FSComm is actually an application that runs an embedded copy of the FreeSwitch core.
> 
> Best to run fs_cli on both the box running FSComm and the freeswitch server and capture debug logs so we can more properly assist in diagnosing your problem.
> 
> -- W



From msc at freeswitch.org  Wed Mar 17 22:52:25 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 22:52:25 -0700
Subject: [Freeswitch-dev] Woohoo! We have call recording working!
In-Reply-To: <4BA11F85.8090803@marketelsystems.com>
References: <4BA11F85.8090803@marketelsystems.com>
Message-ID: <87f2f3b91003172252u1096d350h833be1a6fef0a439@mail.gmail.com>

On Wed, Mar 17, 2010 at 11:29 AM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

> As per Andrew Thompson's suggestion, I changed the processing of the
> call recording to start recording as part of processing a CHANNEL_BRIDGE
> event.  It turns out that (at least with Erlang) the call is not
> actually bridged when the api command to bridge a call returns ok; the
> ok only means the bridge command was successfully queued for processing,
> not that it's completed.
>
> I guess I wouldn't call this a bug with call recording, but it's
> definitely a "gotcha".
>

Nope, not a bug - that's by design. The transmission of the CHANNEL_BRIDGE
event means it was successful. The OK+ from the API call to bridge simply
means that the bridge command was successfully submitted for processing, not
that the processing of it was successful. This is the power, beauty, and
elegance of the event system in FS. Events tell you pretty much *everything*
that is going on and you can pretty much do *anything* when you harness the
power of events.
-MC
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From vetali100 at gmail.com  Fri Mar 19 03:51:08 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 19 Mar 2010 12:51:08 +0200
Subject: [Freeswitch-dev] **/usr/local/freeswitch/mod/mod_spidermonkey.so:
	undefined symbol: SSL_connect**
Message-ID: <a858e7691003190351g28d208baw2f4269214db48539@mail.gmail.com>

Hi,

I've built last trunk today and am getting this error during FreeSWITCH
loading.
   **/usr/local/freeswitch/mod/mod_spidermonkey.so: undefined symbol:
SSL_connect**
Same error for mod_xml_cdr...

Looks like it was fixed as per the following:
http://jira.freeswitch.org/browse/FSBUILD-250

But error exists for me and I am not sure how to resolve it.

*I tried to rebuild again and noticed the following warning (never seen this
before...):*

checking for curl... no
checking for tgetent in -lncurses... yes
checking openssl/tls1.h usability... no
checking openssl/tls1.h presence... yes
configure: WARNING: openssl/tls1.h: present but cannot be compiled
configure: WARNING: openssl/tls1.h:     check for missing prerequisite
headers?
configure: WARNING: openssl/tls1.h: see the Autoconf documentation
configure: WARNING: openssl/tls1.h:     section "Present But Cannot Be
Compiled"
configure: WARNING: openssl/tls1.h: proceeding with the compiler's result
configure: WARNING:     ## --------------------------------- ##
configure: WARNING:     ## Report this to BUG-REPORT-ADDRESS ##
configure: WARNING:     ## --------------------------------- ##
checking for openssl/tls1.h... no
configure: WARNING: OpenSSL include files were not found


*Can this be the reason of error and (newbie question) how can I find
OpenSSL include files?*


Thank you,
vIT
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From vetali100 at gmail.com  Fri Mar 19 04:25:45 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 19 Mar 2010 13:25:45 +0200
Subject: [Freeswitch-dev]
	**/usr/local/freeswitch/mod/mod_spidermonkey.so: undefined
	symbol: SSL_connect**
In-Reply-To: <a858e7691003190351g28d208baw2f4269214db48539@mail.gmail.com>
References: <a858e7691003190351g28d208baw2f4269214db48539@mail.gmail.com>
Message-ID: <a858e7691003190425x1b1f9b0ds4062c08461542f84@mail.gmail.com>

Found the following in config.log

configure:20286: checking openssl/tls1.h usability
configure:20286: gcc -c -g -O2  conftest.c >&5
In file included from conftest.c:127:
/usr/include/openssl/tls1.h:125: error: expected ';', ',' or ')' before '*'
token
/usr/include/openssl/tls1.h:126: error: expected ';', ',' or ')' before '*'
token
configure:20286: $? = 1
configure: failed program was:
| /* confdefs.h */


My tls1.h in /usr/include had the following lines 125,126:
const char *SSL_get_servername(const SSL *s, const int type) ;
int SSL_get_servername_type(const SSL *s) ;
Strange, these particular lines do not look wrong for me.

I found same file in OpenSSL source and put instead of my file.
It gives me the same compilation problem.

Will compile FS without SSL.

Any idea why this happens?


Thank you,
vIT

2010/3/19 Vitalii Colosov <vetali100 at gmail.com>

> Hi,
>
> I've built last trunk today and am getting this error during FreeSWITCH
> loading.
>    **/usr/local/freeswitch/mod/mod_spidermonkey.so: undefined symbol:
> SSL_connect**
> Same error for mod_xml_cdr...
>
> Looks like it was fixed as per the following:
> http://jira.freeswitch.org/browse/FSBUILD-250
>
> But error exists for me and I am not sure how to resolve it.
>
> *I tried to rebuild again and noticed the following warning (never seen
> this before...):*
>
> checking for curl... no
> checking for tgetent in -lncurses... yes
> checking openssl/tls1.h usability... no
> checking openssl/tls1.h presence... yes
> configure: WARNING: openssl/tls1.h: present but cannot be compiled
> configure: WARNING: openssl/tls1.h:     check for missing prerequisite
> headers?
> configure: WARNING: openssl/tls1.h: see the Autoconf documentation
> configure: WARNING: openssl/tls1.h:     section "Present But Cannot Be
> Compiled"
> configure: WARNING: openssl/tls1.h: proceeding with the compiler's result
> configure: WARNING:     ## --------------------------------- ##
> configure: WARNING:     ## Report this to BUG-REPORT-ADDRESS ##
> configure: WARNING:     ## --------------------------------- ##
> checking for openssl/tls1.h... no
> configure: WARNING: OpenSSL include files were not found
>
>
> *Can this be the reason of error and (newbie question) how can I find
> OpenSSL include files?*
>
>
> Thank you,
> vIT
>
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From brian at freeswitch.org  Fri Mar 19 06:23:46 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Mar 2010 08:23:46 -0500
Subject: [Freeswitch-dev]
	**/usr/local/freeswitch/mod/mod_spidermonkey.so: undefined
	symbol: SSL_connect**
In-Reply-To: <a858e7691003190351g28d208baw2f4269214db48539@mail.gmail.com>
References: <a858e7691003190351g28d208baw2f4269214db48539@mail.gmail.com>
Message-ID: <BA3CA902-2D9F-407E-AFD0-A9435D505E35@freeswitch.org>

Sounds like you are missing the OpenSSL devel package.

/b

On Mar 19, 2010, at 5:51 AM, Vitalii Colosov wrote:

> Hi,
> 
> I've built last trunk today and am getting this error during FreeSWITCH loading.
>    **/usr/local/freeswitch/mod/mod_spidermonkey.so: undefined symbol: SSL_connect** 
> Same error for mod_xml_cdr...

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From mike at jerris.com  Fri Mar 19 06:24:19 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Mar 2010 09:24:19 -0400
Subject: [Freeswitch-dev]
	**/usr/local/freeswitch/mod/mod_spidermonkey.so: undefined
	symbol: SSL_connect**
In-Reply-To: <a858e7691003190425x1b1f9b0ds4062c08461542f84@mail.gmail.com>
References: <a858e7691003190351g28d208baw2f4269214db48539@mail.gmail.com>
	<a858e7691003190425x1b1f9b0ds4062c08461542f84@mail.gmail.com>
Message-ID: <15EAE2DF-B389-4E24-8CB1-ECF780B97AD8@jerris.com>

SSL is an undefined type for whatever reason.  It sounds like some sort of issue with your ssl install or conflicting stuff installed to your includes
On Mar 19, 2010, at 7:25 AM, Vitalii Colosov wrote:

> Found the following in config.log
> 
> configure:20286: checking openssl/tls1.h usability
> configure:20286: gcc -c -g -O2  conftest.c >&5
> In file included from conftest.c:127:
> /usr/include/openssl/tls1.h:125: error: expected ';', ',' or ')' before '*' token
> /usr/include/openssl/tls1.h:126: error: expected ';', ',' or ')' before '*' token
> configure:20286: $? = 1
> configure: failed program was:
> | /* confdefs.h */
> 
> 
> My tls1.h in /usr/include had the following lines 125,126:
> const char *SSL_get_servername(const SSL *s, const int type) ;
> int SSL_get_servername_type(const SSL *s) ;
> Strange, these particular lines do not look wrong for me.
> 
> I found same file in OpenSSL source and put instead of my file.
> It gives me the same compilation problem.
> 
> Will compile FS without SSL. 
> 
> Any idea why this happens?
> 
> 
> Thank you,
> vIT
> 
> 2010/3/19 Vitalii Colosov <vetali100 at gmail.com>
> Hi,
> 
> I've built last trunk today and am getting this error during FreeSWITCH loading.
>    **/usr/local/freeswitch/mod/mod_spidermonkey.so: undefined symbol: SSL_connect** 
> Same error for mod_xml_cdr...
> 
> Looks like it was fixed as per the following: http://jira.freeswitch.org/browse/FSBUILD-250
> 
> But error exists for me and I am not sure how to resolve it.
> 
> I tried to rebuild again and noticed the following warning (never seen this before...):
> 
> checking for curl... no
> checking for tgetent in -lncurses... yes
> checking openssl/tls1.h usability... no
> checking openssl/tls1.h presence... yes
> configure: WARNING: openssl/tls1.h: present but cannot be compiled
> configure: WARNING: openssl/tls1.h:     check for missing prerequisite headers?
> configure: WARNING: openssl/tls1.h: see the Autoconf documentation
> configure: WARNING: openssl/tls1.h:     section "Present But Cannot Be Compiled"
> configure: WARNING: openssl/tls1.h: proceeding with the compiler's result
> configure: WARNING:     ## --------------------------------- ##
> configure: WARNING:     ## Report this to BUG-REPORT-ADDRESS ##
> configure: WARNING:     ## --------------------------------- ##
> checking for openssl/tls1.h... no
> configure: WARNING: OpenSSL include files were not found
> 
> 
> Can this be the reason of error and (newbie question) how can I find OpenSSL include files?
> 
> 
> Thank you,
> vIT
> 
> _______________________________________________
> 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 Suneel.Papineni at mettoni.com  Fri Mar 19 08:20:06 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 19 Mar 2010 15:20:06 -0000
Subject: [Freeswitch-dev] Conference with FSComm
Message-ID: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>

Hi,

 

I am trying to achieve 3-way (or 3-party) conference scenario using
FSComm. Scenario is as follows:

 

An incoming call comes to FSComm from an extension 'A' and is answered.
Now FSComm makes a call to another extension 'B' and is answered. Then I
want to bridge all three calls and be in conference. So all three
parties A, B and FSComm will be in conference.

 

Could someone please give me an idea how can I achieve this scenario
using Event Sockets. (want to test with fs_cli using commands).

 

Thanks & Regards

Suneel

 


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
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are addressed. If you have received this email in error please notify
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From m.sobkow at marketelsystems.com  Fri Mar 19 09:49:15 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 19 Mar 2010 10:49:15 -0600
Subject: [Freeswitch-dev] Woohoo!  I'm done core coding!
Message-ID: <4BA3AB0B.9040800@marketelsystems.com>

Just thought I'd let you all know that I finished the core coding for 
our project ahead of schedule, thanks to Andrew's help.  All that's left 
is testing, bug fixes, and client-side work for our applications.  The 
Freeswitch integration core is done, done, done!  Ah feel gud.  Ah nu 
that ah wud!  Ah feelz nice, like sugah an' spice! :D

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From dave at 3c.co.uk  Fri Mar 19 09:56:44 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 19 Mar 2010 10:56:44 -0600
Subject: [Freeswitch-dev] Woohoo!  I'm done core coding!
In-Reply-To: <4BA3AB0B.9040800@marketelsystems.com>
References: <4BA3AB0B.9040800@marketelsystems.com>
Message-ID: <1269017804.31795.61.camel@local.freepabx.com>


> Just thought I'd let you all know that I finished the core coding for 
> our project ahead of schedule, thanks to Andrew's help.  All that's left 
> is testing, bug fixes, and client-side work for our applications.  The 
> Freeswitch integration core is done, done, done!  Ah feel gud.  Ah nu 
> that ah wud!  Ah feelz nice, like sugah an' spice! :D

Sounds to me like you need to get out from behind that monitor ;-)
Congratulations, and have a top weekend.

--Dave



From anthony.minessale at gmail.com  Fri Mar 19 10:04:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:04:58 -0500
Subject: [Freeswitch-dev] Woohoo! I'm done core coding!
In-Reply-To: <4BA3AB0B.9040800@marketelsystems.com>
References: <4BA3AB0B.9040800@marketelsystems.com>
Message-ID: <191c3a031003191004k50642c98p2a01840ab0af19b1@mail.gmail.com>

Now you have more fun things to think about like ClueCon MMX!

On Fri, Mar 19, 2010 at 11:49 AM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

> Just thought I'd let you all know that I finished the core coding for
> our project ahead of schedule, thanks to Andrew's help.  All that's left
> is testing, bug fixes, and client-side work for our applications.  The
> Freeswitch integration core is done, done, done!  Ah feel gud.  Ah nu
> that ah wud!  Ah feelz nice, like sugah an' spice! :D
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From Suneel.Papineni at mettoni.com  Fri Mar 19 11:20:44 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 19 Mar 2010 18:20:44 -0000
Subject: [Freeswitch-dev]  Conference with FSComm
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>
Message-ID: <3181A30B8C35AB4AA8577B78DDF46138069F81D9@nickel.mettonigroup.com>

I am not sure if the following method works. Please let me know how
could I implement this through event sockets.

 

'A' calls FSComm and FSComm calls 'B'. Now FSComm is having call details
of all call legs. If possible FSComm now creates a conference and join
all 3 call legs to the conference. This might solve the problem, but
could you please let me know how can create a conference and join all
the legs. Please let me know the commands for this. I checked with
"mod_conference" module, unable to find respective commands. I tried
"api conference....", it is saying command not found. I tried
"conference....." command, it is also not working.

 

Thanks & Regards

Suneel

 

 

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Suneel
Papineni
Sent: 19 March 2010 15:20
To: freeswitch-dev at lists.freeswitch.org
Cc: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-dev] Conference with FSComm

 

Hi,

 

I am trying to achieve 3-way (or 3-party) conference scenario using
FSComm. Scenario is as follows:

 

An incoming call comes to FSComm from an extension 'A' and is answered.
Now FSComm makes a call to another extension 'B' and is answered. Then I
want to bridge all three calls and be in conference. So all three
parties A, B and FSComm will be in conference.

 

Could someone please give me an idea how can I achieve this scenario
using Event Sockets. (want to test with fs_cli using commands).

 

Thanks & Regards

Suneel

 

************************************************************************
*
Please consider the environment before printing this e-mail
************************************************************************
*
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com
 
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Registered in England and Wales: 4485956
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*************************************************************************
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From scaram at hotmail.de  Mon Mar 22 03:33:30 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Mon, 22 Mar 2010 11:33:30 +0100
Subject: [Freeswitch-dev] hooked functions are not called while
	using	one single session
In-Reply-To: <191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>,
	<191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>,
	<SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>,
	<191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>
Message-ID: <SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>


I am now using the application interface and play a file after answering the call. The call is not connected to a another session, it's just a single session.

SWITCH_STANDARD_APP(mod_my_function)
{
     /* play audio file */
    switch_ivr_play_file(session, NULL, "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav", NULL);
}

Now the hooks for channel_write_frame and channel_read_frame are called! I write the audio frames into a file for testing if everything is correct. The data written in channel_write_frame seems to be perfect, but audio written in channel_read_frame only seems to be good at first sight. If I record 20 seconds of audio and listen to the audiofile I hear that 2 or 3 times short parts are missing. 

static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
  // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into a file
  fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t **frame, switch_io_flag_t flags, int stream_id)
{
    // Spoken audio samples from my microphone are written, but sometimes samples are missing
    fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}

Is it possible that I don't get all audio samples by hooked functions? why audio recorded in channel_write_frame is brilliant and in channel_read_frame not?






Date: Tue, 16 Mar 2010 13:59:28 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you probably want to use the application interface and a while loop while reading a frame.

see most of the functions in switch_ivr_play_say


On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:






The module I am writing should be an endpoint VOIP-interface for another program. The module should answer a call and transport incoming audio into a 3rd party application. Furthermore the 3rd party application sends audio data back to the caller. This is why I hooked channel_read_frame and channel_write_frame. 


How can I trigger the channel to do something that requires reading or writing audio? Does it make a difference to use the endpoint interface instead of the hooks? 


Date: Tue, 16 Mar 2010 08:34:18 -0500

From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session


What exactly is your goal?

those hooks will probably only be called if you send the channel to do something that 
requires reading and writing audio.


On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:







Hello,
I am writing my own freeswitch module and have implemented 2 hooks for reading and writing audio (I did not implement the endpoint interface).
My problem is that the channel_write_frame and channel_read_frame callback-functions are only called if I make an intercept with another session (switch_ivr_intercept_session), but my usecase has only one single Session for getting and putting audio data in my module. 



How can I achieve this? 
scaram

/* on channel init I answer the session and set hooks */
static switch_status_t channel_on_init(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);


    switch_channel_answer(channel);

    switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
    switch_set_flag(tech_pvt, TFLAG_IO);

    switch_core_event_hook_add_write_frame(session, &channel_write_frame);


    switch_core_event_hook_add_read_frame(session, &channel_read_frame);
}

static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)


{
    /* is only called when I intercept 2 Sessions */
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
    /* is only called when I intercept 2 Sessions */


}


 		 	   		  
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-- 
Anthony Minessale II

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


Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com

GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

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sip:888 at conference.freeswitch.org

iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
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Alles in einem Postfach  ?   Ich will Hotmail!


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http://www.freeswitch.org




-- 
Anthony Minessale II

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

Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
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googletalk:conf+888 at conference.freeswitch.org
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From anthony.minessale at gmail.com  Mon Mar 22 09:55:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Mar 2010 11:55:42 -0500
Subject: [Freeswitch-dev] hooked functions are not called while using
	one single session
In-Reply-To: <SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>
	<191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>
	<SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>
	<191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>
	<SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>
Message-ID: <191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>

Why do you even need the hooks?
What exactly are you trying to do?
Maybe you should explain it.


On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:

>  I am now using the application interface and play a file after answering
> the call. The call is not connected to a another session, it's just a single
> session.
>
> SWITCH_STANDARD_APP(mod_my_function)
> {
>      /* play audio file */
>     switch_ivr_play_file(session, NULL,
> "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav",
> NULL);
> }
>
> Now the hooks for channel_write_frame and channel_read_frame are called! I
> write the audio frames into a file for testing if everything is correct. The
> data written in channel_write_frame seems to be perfect, but audio written
> in channel_read_frame only seems to be good at first sight. If I record 20
> seconds of audio and listen to the audiofile I hear that 2 or 3 times short
> parts are missing.
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>   // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into
> a file
>   fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
>
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t **frame, switch_io_flag_t flags, int stream_id)
> {
>     // Spoken audio samples from my microphone are written, but sometimes
> samples are missing
>     fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
> }
>
> Is it possible that I don't get all audio samples by hooked functions? why
> audio recorded in channel_write_frame is brilliant and in channel_read_frame
> not?
>
>
>
>
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 13:59:28 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you probably want to use the application interface and a while loop while
> reading a frame.
>
> see most of the functions in switch_ivr_play_say
>
>
> On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  The module I am writing should be an endpoint VOIP-interface for another
> program. The module should answer a call and transport incoming audio into a
> 3rd party application. Furthermore the 3rd party application sends audio
> data back to the caller. This is why I hooked channel_read_frame and
> channel_write_frame.
>
> How can I trigger the channel to do something that requires reading or
> writing audio? Does it make a difference to use the endpoint interface
> instead of the hooks?
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 08:34:18 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
>
> What exactly is your goal?
>
> those hooks will probably only be called if you send the channel to do
> something that
> requires reading and writing audio.
>
>
> On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  Hello,
> I am writing my own freeswitch module and have implemented 2 hooks for
> reading and writing audio (I did not implement the endpoint interface).
> My problem is that the channel_write_frame and channel_read_frame
> callback-functions are only called if I make an intercept with another
> session (switch_ivr_intercept_session), but my usecase has only one single
> Session for getting and putting audio data in my module.
>
> How can I achieve this?
> scaram
>
> /* on channel init I answer the session and set hooks */
> static switch_status_t channel_on_init(switch_core_session_t *session)
> {
>     switch_channel_t *channel = switch_core_session_get_channel(session);
>     switch_channel_answer(channel);
>
>     switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
>     switch_set_flag(tech_pvt, TFLAG_IO);
>
>     switch_core_event_hook_add_write_frame(session, &channel_write_frame);
>     switch_core_event_hook_add_read_frame(session, &channel_read_frame);
> }
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
>
>
>
> ------------------------------
> Ein Postfach f?r Alles ?   bei Hotmail 5 E-Mail-Adressen online verwalten!<http://redirect.gimas.net/?n=M1003HM5Adressen>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
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> 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
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
> 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
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Treffe Freunde im Messenger Videochat!<http://redirect.gimas.net/?n=M1003IMVideochat>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Mon Mar 22 11:08:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Mar 2010 13:08:34 -0500
Subject: [Freeswitch-dev] Now Hiring for CudaTEL UI Programmer
Message-ID: <191c3a031003221108u1c502219qef68969a6656229@mail.gmail.com>

As most people know, the CudaTEL uses FreeSWITCH internally as the telephony
engine.
We're now looking to expand the development team:

We're looking for a UI programmer with jquery experience.

See:

http://jobs.jsninja.com/job/1598/jqueryjs-ui-developer-at-barracuda-networks/

Anyone interested,  please apply to this forum or email jobs at freeswitch.org

Thanks.


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From scaram at hotmail.de  Mon Mar 22 16:23:50 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Tue, 23 Mar 2010 00:23:50 +0100
Subject: [Freeswitch-dev] hooked functions are not called while
	using	one single session
In-Reply-To: <191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>,
	<191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>,
	<SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>,
	<191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>,
	<SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>,
	<191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>
Message-ID: <SNT133-w324431DD601582973573FEA0270@phx.gbl>


I have an selfmade software which is controlled by a telephone call (ISDN/CAPI). The user calls with his phone, gets 
connected and can speak commands which are detected by an internal speech recognizer. Furthermore the user can accept 
or hangup incoming calls just by his voice command. It is also possible to dial out. 

Now I am adding VOIP functionality to my system and decided to use FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP interface for my software. The module must be able to:

1. recognize an incoming call which is the users control session (single session)
   I do this in statehandler function channel_on_init.
2. pick up another incoming call and intercept with my users control session. 
3. Get the audio data from voip session (user control session) and send it to my application (by own tcp-socket)
    My idea is to use channel_write_frame to receive audio.
3. Send audio data from my application to voip user control session so the user can hear audio.
    My idea is to use channel_read_frame.

explanation for using hooks:
----------------------------
The endpoint interface callback functions where never called in my module, even I tried to adapt the way it is done 
in mod_portaudio. So my workaround was to set hook functions channel_read_frame/channel_write_frame to handle audio. 

My problems:
------------
The hook functions are working fine if I have 2 sessions (put together by switch_ivr_intercept_session), but if I only have the single user control session no hooked function is called. In this case the only way to achieve that hooks are called, is to play audio with switch_ivr_play_file. But then, audio samples got lost sometimes.

Maybe there is a better way to solve my requirements? 




Date: Mon, 22 Mar 2010 11:55:42 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

Why do you even need the hooks?
What exactly are you trying to do?
Maybe you should explain it.


On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:






I am now using the application interface and play a file after answering the call. The call is not connected to a another session, it's just a single session.

SWITCH_STANDARD_APP(mod_my_function)
{
     /* play audio file */

    switch_ivr_play_file(session, NULL, "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav", NULL);
}

Now the hooks for channel_write_frame and channel_read_frame are called! I write the audio frames into a file for testing if everything is correct. The data written in channel_write_frame seems to be perfect, but audio written in channel_read_frame only seems to be good at first sight. If I record 20 seconds of audio and listen to the audiofile I hear that 2 or 3 times short parts are missing. 


static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
  // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into a file

  fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t **frame, switch_io_flag_t flags, int stream_id)

{
    // Spoken audio samples from my microphone are written, but sometimes samples are missing
    fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}

Is it possible that I don't get all audio samples by hooked functions? why audio recorded in channel_write_frame is brilliant and in channel_read_frame not?







Date: Tue, 16 Mar 2010 13:59:28 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org

Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you probably want to use the application interface and a while loop while reading a frame.

see most of the functions in switch_ivr_play_say



On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:






The module I am writing should be an endpoint VOIP-interface for another program. The module should answer a call and transport incoming audio into a 3rd party application. Furthermore the 3rd party application sends audio data back to the caller. This is why I hooked channel_read_frame and channel_write_frame. 



How can I trigger the channel to do something that requires reading or writing audio? Does it make a difference to use the endpoint interface instead of the hooks? 


Date: Tue, 16 Mar 2010 08:34:18 -0500


From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session



What exactly is your goal?

those hooks will probably only be called if you send the channel to do something that 
requires reading and writing audio.


On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:








Hello,
I am writing my own freeswitch module and have implemented 2 hooks for reading and writing audio (I did not implement the endpoint interface).
My problem is that the channel_write_frame and channel_read_frame callback-functions are only called if I make an intercept with another session (switch_ivr_intercept_session), but my usecase has only one single Session for getting and putting audio data in my module. 




How can I achieve this? 
scaram

/* on channel init I answer the session and set hooks */
static switch_status_t channel_on_init(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);



    switch_channel_answer(channel);

    switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
    switch_set_flag(tech_pvt, TFLAG_IO);

    switch_core_event_hook_add_write_frame(session, &channel_write_frame);



    switch_core_event_hook_add_read_frame(session, &channel_read_frame);
}

static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)



{
    /* is only called when I intercept 2 Sessions */
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
    /* is only called when I intercept 2 Sessions */



}


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



Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com


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IRC: irc.freenode.net #freeswitch

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Alles in einem Postfach  ?   Ich will Hotmail!



_______________________________________________

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev

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http://www.freeswitch.org




-- 
Anthony Minessale II

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


Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com

GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org

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-- 
Anthony Minessale II

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

Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
 		 	   		  
_________________________________________________________________
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From anthony.minessale at gmail.com  Mon Mar 22 16:34:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Mar 2010 18:34:46 -0500
Subject: [Freeswitch-dev] hooked functions are not called while using
	one single session
In-Reply-To: <SNT133-w324431DD601582973573FEA0270@phx.gbl>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>
	<191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>
	<SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>
	<191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>
	<SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>
	<191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>
	<SNT133-w324431DD601582973573FEA0270@phx.gbl>
Message-ID: <191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>

you don't use the hook functions you use a frame pointer

switch_frame_t *read_frame;

switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE,
0);

when i said to look at the stream file example i meant in the function
itself to see how
to read and write audio from a channel in a foreground loop.



On Mon, Mar 22, 2010 at 6:23 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:

>  I have an selfmade software which is controlled by a telephone call
> (ISDN/CAPI). The user calls with his phone, gets
> connected and can speak commands which are detected by an internal speech
> recognizer. Furthermore the user can accept
> or hangup incoming calls just by his voice command. It is also possible to
> dial out.
>
> Now I am adding VOIP functionality to my system and decided to use
> FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP
> interface for my software. The module must be able to:
>
> 1. recognize an incoming call which is the users control session (single
> session)
>    I do this in statehandler function channel_on_init.
> 2. pick up another incoming call and intercept with my users control
> session.
> 3. Get the audio data from voip session (user control session) and send it
> to my application (by own tcp-socket)
>     My idea is to use channel_write_frame to receive audio.
> 3. Send audio data from my application to voip user control session so the
> user can hear audio.
>     My idea is to use channel_read_frame.
>
> explanation for using hooks:
> ----------------------------
> The endpoint interface callback functions where never called in my module,
> even I tried to adapt the way it is done
> in mod_portaudio. So my workaround was to set hook functions
> channel_read_frame/channel_write_frame to handle audio.
>
> My problems:
> ------------
> The hook functions are working fine if I have 2 sessions (put together by
> switch_ivr_intercept_session), but if I only have the single user control
> session no hooked function is called. In this case the only way to achieve
> that hooks are called, is to play audio with switch_ivr_play_file. But then,
> audio samples got lost sometimes.
>
> Maybe there is a better way to solve my requirements?
>
>
>
>
> ------------------------------
> Date: Mon, 22 Mar 2010 11:55:42 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> Why do you even need the hooks?
> What exactly are you trying to do?
> Maybe you should explain it.
>
>
> On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  I am now using the application interface and play a file after answering
> the call. The call is not connected to a another session, it's just a single
> session.
>
> SWITCH_STANDARD_APP(mod_my_function)
> {
>      /* play audio file */
>     switch_ivr_play_file(session, NULL,
> "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav",
> NULL);
> }
>
> Now the hooks for channel_write_frame and channel_read_frame are called! I
> write the audio frames into a file for testing if everything is correct. The
> data written in channel_write_frame seems to be perfect, but audio written
> in channel_read_frame only seems to be good at first sight. If I record 20
> seconds of audio and listen to the audiofile I hear that 2 or 3 times short
> parts are missing.
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>   // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into
> a file
>   fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
>
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t **frame, switch_io_flag_t flags, int stream_id)
> {
>     // Spoken audio samples from my microphone are written, but sometimes
> samples are missing
>     fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
> }
>
> Is it possible that I don't get all audio samples by hooked functions? why
> audio recorded in channel_write_frame is brilliant and in channel_read_frame
> not?
>
>
>
>
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 13:59:28 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you probably want to use the application interface and a while loop while
> reading a frame.
>
> see most of the functions in switch_ivr_play_say
>
>
> On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  The module I am writing should be an endpoint VOIP-interface for another
> program. The module should answer a call and transport incoming audio into a
> 3rd party application. Furthermore the 3rd party application sends audio
> data back to the caller. This is why I hooked channel_read_frame and
> channel_write_frame.
>
> How can I trigger the channel to do something that requires reading or
> writing audio? Does it make a difference to use the endpoint interface
> instead of the hooks?
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 08:34:18 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
>
> What exactly is your goal?
>
> those hooks will probably only be called if you send the channel to do
> something that
> requires reading and writing audio.
>
>
> On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  Hello,
> I am writing my own freeswitch module and have implemented 2 hooks for
> reading and writing audio (I did not implement the endpoint interface).
> My problem is that the channel_write_frame and channel_read_frame
> callback-functions are only called if I make an intercept with another
> session (switch_ivr_intercept_session), but my usecase has only one single
> Session for getting and putting audio data in my module.
>
> How can I achieve this?
> scaram
>
> /* on channel init I answer the session and set hooks */
> static switch_status_t channel_on_init(switch_core_session_t *session)
> {
>     switch_channel_t *channel = switch_core_session_get_channel(session);
>     switch_channel_answer(channel);
>
>     switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
>     switch_set_flag(tech_pvt, TFLAG_IO);
>
>     switch_core_event_hook_add_write_frame(session, &channel_write_frame);
>     switch_core_event_hook_add_read_frame(session, &channel_read_frame);
> }
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
>
>
>
> ------------------------------
> Ein Postfach f?r Alles ?   bei Hotmail 5 E-Mail-Adressen online verwalten!<http://redirect.gimas.net/?n=M1003HM5Adressen>
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> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
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> --
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>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
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>
> _______________________________________________
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> http://www.freeswitch.org
>
>
>
>
> --
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>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
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>
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> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
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> 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 <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From cstomi.levlist at gmail.com  Tue Mar 23 04:38:10 2010
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Tue, 23 Mar 2010 12:38:10 +0100
Subject: [Freeswitch-dev] windows json build
In-Reply-To: <E1NtiY8-0005qO-65@svn.freeswitch.org>
References: <E1NtiY8-0005qO-65@svn.freeswitch.org>
Message-ID: <4BA8A822.4020401@gmail.com>

Hello,

I've seen the json upgrade (0.8 ->0.9) has been done for windows as well 
in r17064.

If it would work, I'd like to fix the windows build for mod_json_cdr
(http://jira.freeswitch.org/browse/MODEVENT-62)

I was trying with vc++ 2008 express, but I can't compile jsonlib.

json_util.c(62) : error C2065: 'O_RDONLY' : undeclared identifier
json_util.c(97) : error C2065: 'O_WRONLY' : undeclared identifier
json_util.c(97) : error C2065: 'O_TRUNC' : undeclared identifier
json_util.c(97) : error C2065: 'O_CREAT' : undeclared identifier

Is it works for you? does mod_curl build works on windows?
or the jsonlib doesn't support windows at all (it seems to me) ?

Thanks in advance,
Tamas




From cstomi.levlist at gmail.com  Tue Mar 23 09:12:23 2010
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Tue, 23 Mar 2010 17:12:23 +0100
Subject: [Freeswitch-dev] windows json build
In-Reply-To: <4BA8A822.4020401@gmail.com>
References: <E1NtiY8-0005qO-65@svn.freeswitch.org> <4BA8A822.4020401@gmail.com>
Message-ID: <4BA8E867.1020702@gmail.com>

Hello,

Responding (for others who may be intrested)
for json build a patched json tarball is needed.

Here is the modification
http://pastebin.freeswitch.org/12524

After this libjson build works.
mod_curl still used json header files from 0.8 version.
And I got compile errors
http://pastebin.freeswitch.org/12525

Jlenk is working on mod_curl build...it's coming soon...
Thanks the help!

Tamas



Tamas Cseke wrote:
> Hello,
>
> I've seen the json upgrade (0.8 ->0.9) has been done for windows as well 
> in r17064.
>
> If it would work, I'd like to fix the windows build for mod_json_cdr
> (http://jira.freeswitch.org/browse/MODEVENT-62)
>
> I was trying with vc++ 2008 express, but I can't compile jsonlib.
>
> json_util.c(62) : error C2065: 'O_RDONLY' : undeclared identifier
> json_util.c(97) : error C2065: 'O_WRONLY' : undeclared identifier
> json_util.c(97) : error C2065: 'O_TRUNC' : undeclared identifier
> json_util.c(97) : error C2065: 'O_CREAT' : undeclared identifier
>
> Is it works for you? does mod_curl build works on windows?
> or the jsonlib doesn't support windows at all (it seems to me) ?
>
> Thanks in advance,
> Tamas
>
>
>
> _______________________________________________
> 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 brian at freeswitch.org  Tue Mar 23 09:15:39 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 11:15:39 -0500
Subject: [Freeswitch-dev] windows json build
In-Reply-To: <4BA8E867.1020702@gmail.com>
References: <E1NtiY8-0005qO-65@svn.freeswitch.org>
	<4BA8A822.4020401@gmail.com> <4BA8E867.1020702@gmail.com>
Message-ID: <F0FDE688-5E55-42DF-ABEA-B8FC53D5BD06@freeswitch.org>

I thought this was fixed already yesterday... did you redownload the json lib?

/b

On Mar 23, 2010, at 11:12 AM, Tamas Cseke wrote:

> Hello,
> 
> Responding (for others who may be intrested)
> for json build a patched json tarball is needed.
> 
> Here is the modification
> http://pastebin.freeswitch.org/12524
> 
> After this libjson build works.
> mod_curl still used json header files from 0.8 version.
> And I got compile errors
> http://pastebin.freeswitch.org/12525
> 
> Jlenk is working on mod_curl build...it's coming soon...
> Thanks the help!
> 
> Tamas
> 
> 
> 
> Tamas Cseke wrote:
>> Hello,
>> 
>> I've seen the json upgrade (0.8 ->0.9) has been done for windows as well 
>> in r17064.
>> 
>> If it would work, I'd like to fix the windows build for mod_json_cdr
>> (http://jira.freeswitch.org/browse/MODEVENT-62)
>> 
>> I was trying with vc++ 2008 express, but I can't compile jsonlib.
>> 
>> json_util.c(62) : error C2065: 'O_RDONLY' : undeclared identifier
>> json_util.c(97) : error C2065: 'O_WRONLY' : undeclared identifier
>> json_util.c(97) : error C2065: 'O_TRUNC' : undeclared identifier
>> json_util.c(97) : error C2065: 'O_CREAT' : undeclared identifier
>> 
>> Is it works for you? does mod_curl build works on windows?
>> or the jsonlib doesn't support windows at all (it seems to me) ?
>> 
>> Thanks in advance,
>> Tamas
>> 
>> 
>> 
>> _______________________________________________
>> 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 mike at jerris.com  Tue Mar 23 09:48:01 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 23 Mar 2010 12:48:01 -0400
Subject: [Freeswitch-dev] windows json build
In-Reply-To: <4BA8E867.1020702@gmail.com>
References: <E1NtiY8-0005qO-65@svn.freeswitch.org>
	<4BA8A822.4020401@gmail.com> <4BA8E867.1020702@gmail.com>
Message-ID: <66E9C220-86E3-4CCF-AD50-83715B5E6AD1@jerris.com>

The new tarball will be up shortly, there was one more change.

Mike

On Mar 23, 2010, at 12:12 PM, Tamas Cseke wrote:

> Hello,
> 
> Responding (for others who may be intrested)
> for json build a patched json tarball is needed.
> 
> Here is the modification
> http://pastebin.freeswitch.org/12524
> 
> After this libjson build works.
> mod_curl still used json header files from 0.8 version.
> And I got compile errors
> http://pastebin.freeswitch.org/12525
> 
> Jlenk is working on mod_curl build...it's coming soon...
> Thanks the help!
> 
> Tamas
> 
> 
> 
> Tamas Cseke wrote:
>> Hello,
>> 
>> I've seen the json upgrade (0.8 ->0.9) has been done for windows as well 
>> in r17064.
>> 
>> If it would work, I'd like to fix the windows build for mod_json_cdr
>> (http://jira.freeswitch.org/browse/MODEVENT-62)
>> 
>> I was trying with vc++ 2008 express, but I can't compile jsonlib.
>> 
>> json_util.c(62) : error C2065: 'O_RDONLY' : undeclared identifier
>> json_util.c(97) : error C2065: 'O_WRONLY' : undeclared identifier
>> json_util.c(97) : error C2065: 'O_TRUNC' : undeclared identifier
>> json_util.c(97) : error C2065: 'O_CREAT' : undeclared identifier
>> 
>> Is it works for you? does mod_curl build works on windows?
>> or the jsonlib doesn't support windows at all (it seems to me) ?
>> 
>> Thanks in advance,
>> Tamas
>> 
>> 
>> 
>> _______________________________________________
>> 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 Suneel.Papineni at mettoni.com  Thu Mar 25 03:47:46 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Thu, 25 Mar 2010 10:47:46 -0000
Subject: [Freeswitch-dev] Core-UUID
Message-ID: <3181A30B8C35AB4AA8577B78DDF4613806A5E4DA@nickel.mettonigroup.com>

Hi,

 

May I know what is the use of Core-UUID in the events. For me (using
FSComm) I am getting same Core-UUID value for all events even if there
are multiple calls.

 

Core-UUID: 3f3e9290-077e-4fc8-80a5-2ee74389b84c

 

Is there any way to distinguish calls (each call might have two call
legs) based on UUID. Each call leg is having unique identifier as
"Unique-ID", but to relate different call leg events to a particular
call, is there any identifier?

 

Thanks & Regards

Suneel

 


*************************************************************************
Please consider the environment before printing this e-mail
*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
the system manager.  http://www.mettoni.com

Mettoni Ltd
Registered in England and Wales: 4485956
9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
*************************************************************************

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From mike at jerris.com  Thu Mar 25 06:11:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 25 Mar 2010 09:11:07 -0400
Subject: [Freeswitch-dev] Core-UUID
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613806A5E4DA@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF4613806A5E4DA@nickel.mettonigroup.com>
Message-ID: <7B169138-26E5-4778-A375-906A83BCAD11@jerris.com>

it relates to the instance for FreeSWITCH.  I cant recall if that one persists reboot or not.

Mike

On Mar 25, 2010, at 6:47 AM, Suneel Papineni wrote:

> Hi,
>  
> May I know what is the use of Core-UUID in the events. For me (using FSComm) I am getting same Core-UUID value for all events even if there are multiple calls.
>  
> Core-UUID: 3f3e9290-077e-4fc8-80a5-2ee74389b84c
>  
> Is there any way to distinguish calls (each call might have two call legs) based on UUID. Each call leg is having unique identifier as ?Unique-ID?, but to relate different call leg events to a particular call, is there any identifier?
>  
> Thanks & Regards
> Suneel
>  
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
> 

Why are you sending confidential emails to a public mailing list?  Should I contact your system manager ?
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN

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From anthony.minessale at gmail.com  Thu Mar 25 06:11:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Mar 2010 08:11:51 -0500
Subject: [Freeswitch-dev] Core-UUID
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613806A5E4DA@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF4613806A5E4DA@nickel.mettonigroup.com>
Message-ID: <191c3a031003250611h68fe5cebpd53e959d20544bc3@mail.gmail.com>

unique per running instance of FS, it changes each time you restart.


On Thu, Mar 25, 2010 at 5:47 AM, Suneel Papineni <
Suneel.Papineni at mettoni.com> wrote:

>  Hi,
>
>
>
> May I know what is the use of Core-UUID in the events. For me (using
> FSComm) I am getting same Core-UUID value for all events even if there are
> multiple calls.
>
>
>
> Core-UUID: 3f3e9290-077e-4fc8-80a5-2ee74389b84c
>
>
>
> Is there any way to distinguish calls (each call might have two call legs)
> based on UUID. Each call leg is having unique identifier as ?Unique-ID?, but
> to relate different call leg events to a particular call, is there any
> identifier?
>
>
>
> Thanks & Regards
>
> Suneel
>
>
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Mar 25 06:28:55 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Mar 2010 08:28:55 -0500
Subject: [Freeswitch-dev] Core-UUID
In-Reply-To: <7B169138-26E5-4778-A375-906A83BCAD11@jerris.com>
References: <3181A30B8C35AB4AA8577B78DDF4613806A5E4DA@nickel.mettonigroup.com>
	<7B169138-26E5-4778-A375-906A83BCAD11@jerris.com>
Message-ID: <191c3a031003250628l7c0e0275xd2ffac8fae20b35d@mail.gmail.com>

The uuid for the calls are called

Unique-ID

and they only pertain to events that are on a channel


On Thu, Mar 25, 2010 at 8:11 AM, Michael Jerris <mike at jerris.com> wrote:

> it relates to the instance for FreeSWITCH.  I cant recall if that one
> persists reboot or not.
>
> Mike
>
> On Mar 25, 2010, at 6:47 AM, Suneel Papineni wrote:
>
> Hi,
>
> May I know what is the use of Core-UUID in the events. For me (using
> FSComm) I am getting same Core-UUID value for all events even if there are
> multiple calls.
>
> Core-UUID: 3f3e9290-077e-4fc8-80a5-2ee74389b84c
>
> Is there any way to distinguish calls (each call might have two call legs)
> based on UUID. Each call leg is having unique identifier as ?Unique-ID?, but
> to relate different call leg events to a particular call, is there any
> identifier?
>
> Thanks & Regards
> Suneel
>
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
>
> Why are you sending confidential emails to a public mailing list?  Should I
> contact your system manager ?
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
>
>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From msc at freeswitch.org  Thu Mar 25 09:33:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Mar 2010 09:33:31 -0700
Subject: [Freeswitch-dev] IRC Day!
Message-ID: <87f2f3b91003250933g43da6e5rd275551e71c46f49@mail.gmail.com>

Hello all FreeSWITCHers!

We invite everyone around the world to join us in #freeswitch on
irc.freenode.net. We want everyone to be online today, even if you just
leave your IRC client running in the background. Also, if you don't have an
IRC client handy then go to http://www.freeswitch.org/ and use the web login
located on the right-hand side bar (below the features list). You'll need a
Java-enable browser.

Everyone have a great day!
-Michael
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From scaram at hotmail.de  Fri Mar 26 05:27:33 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Fri, 26 Mar 2010 13:27:33 +0100
Subject: [Freeswitch-dev] hooked functions are not called while
	using	one single session
In-Reply-To: <191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>,
	<191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>,
	<SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>,
	<191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>,
	<SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>,
	<191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>,
	<SNT133-w324431DD601582973573FEA0270@phx.gbl>,
	<191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>
Message-ID: <SNT133-w55A5A011A53DC01B04C610A0230@phx.gbl>


Now I call switch_core_session_read_frame and switch_core_session_write_frame in the same foreground loop. Then I write audio into a file and as a result I see that there are many audio gaps. 

There seems to be a cross effect by using switch_core_session_write_frame and switch_core_session_read_frame together.
If I only use switch_core_session_read_frame in the loop, there are no gaps and everything sounds perfect!

What could be the problem? 



while (1)
{
    // READ
    tstatus = switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);
    if (tstatus==SWITCH_STATUS_SUCCESS)
    {        // write audio in a testfile
            fwrite((short *)(read_frame)->data, 2, ((read_frame)->datalen)/2 ,tmp);
            switch_size_t ts_rf=read_frame->timestamp;
    }

    // WRITE
    vector<unsigned char> vucIn;
    if (getAudio(&vucAudio)) // get next audio and send it back
    {
        memcpy( write_frame.data, &vucIn[0], vucIn.size());
        write_frame.datalen=vucIn.size();
        tstatus = switch_core_session_write_frame(session, &write_frame, SWITCH_IO_FLAG_NONE, 0);
    }
    //Sleep(1);//?
}

Date: Mon, 22 Mar 2010 18:34:46 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you don't use the hook functions you use a frame pointer 

switch_frame_t *read_frame;

switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);

when i said to look at the stream file example i meant in the function itself to see how 

to read and write audio from a channel in a foreground loop.



On Mon, Mar 22, 2010 at 6:23 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:






I have an selfmade software which is controlled by a telephone call (ISDN/CAPI). The user calls with his phone, gets 
connected and can speak commands which are detected by an internal speech recognizer. Furthermore the user can accept 

or hangup incoming calls just by his voice command. It is also possible to dial out. 

Now I am adding VOIP functionality to my system and decided to use FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP interface for my software. The module must be able to:


1. recognize an incoming call which is the users control session (single session)
   I do this in statehandler function channel_on_init.
2. pick up another incoming call and intercept with my users control session. 

3. Get the audio data from voip session (user control session) and send it to my application (by own tcp-socket)
    My idea is to use channel_write_frame to receive audio.
3. Send audio data from my application to voip user control session so the user can hear audio.

    My idea is to use channel_read_frame.

explanation for using hooks:
----------------------------
The endpoint interface callback functions where never called in my module, even I tried to adapt the way it is done 

in mod_portaudio. So my workaround was to set hook functions channel_read_frame/channel_write_frame to handle audio. 

My problems:
------------
The hook functions are working fine if I have 2 sessions (put together by switch_ivr_intercept_session), but if I only have the single user control session no hooked function is called. In this case the only way to achieve that hooks are called, is to play audio with switch_ivr_play_file. But then, audio samples got lost sometimes.


Maybe there is a better way to solve my requirements? 




Date: Mon, 22 Mar 2010 11:55:42 -0500
From: anthony.minessale at gmail.com

To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

Why do you even need the hooks?

What exactly are you trying to do?
Maybe you should explain it.


On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:







I am now using the application interface and play a file after answering the call. The call is not connected to a another session, it's just a single session.

SWITCH_STANDARD_APP(mod_my_function)
{
     /* play audio file */


    switch_ivr_play_file(session, NULL, "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav", NULL);
}

Now the hooks for channel_write_frame and channel_read_frame are called! I write the audio frames into a file for testing if everything is correct. The data written in channel_write_frame seems to be perfect, but audio written in channel_read_frame only seems to be good at first sight. If I record 20 seconds of audio and listen to the audiofile I hear that 2 or 3 times short parts are missing. 



static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
  // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into a file


  fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t **frame, switch_io_flag_t flags, int stream_id)

{
    // Spoken audio samples from my microphone are written, but sometimes samples are missing
    fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}

Is it possible that I don't get all audio samples by hooked functions? why audio recorded in channel_write_frame is brilliant and in channel_read_frame not?








Date: Tue, 16 Mar 2010 13:59:28 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org


Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you probably want to use the application interface and a while loop while reading a frame.

see most of the functions in switch_ivr_play_say




On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:






The module I am writing should be an endpoint VOIP-interface for another program. The module should answer a call and transport incoming audio into a 3rd party application. Furthermore the 3rd party application sends audio data back to the caller. This is why I hooked channel_read_frame and channel_write_frame. 




How can I trigger the channel to do something that requires reading or writing audio? Does it make a difference to use the endpoint interface instead of the hooks? 


Date: Tue, 16 Mar 2010 08:34:18 -0500



From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session




What exactly is your goal?

those hooks will probably only be called if you send the channel to do something that 
requires reading and writing audio.


On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:









Hello,
I am writing my own freeswitch module and have implemented 2 hooks for reading and writing audio (I did not implement the endpoint interface).
My problem is that the channel_write_frame and channel_read_frame callback-functions are only called if I make an intercept with another session (switch_ivr_intercept_session), but my usecase has only one single Session for getting and putting audio data in my module. 





How can I achieve this? 
scaram

/* on channel init I answer the session and set hooks */
static switch_status_t channel_on_init(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);




    switch_channel_answer(channel);

    switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
    switch_set_flag(tech_pvt, TFLAG_IO);

    switch_core_event_hook_add_write_frame(session, &channel_write_frame);




    switch_core_event_hook_add_read_frame(session, &channel_read_frame);
}

static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)




{
    /* is only called when I intercept 2 Sessions */
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
    /* is only called when I intercept 2 Sessions */




}


 		 	   		  
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_______________________________________________

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




Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com



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IRC: irc.freenode.net #freeswitch

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_______________________________________________

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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


iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
 		 	   		  
Treffe Freunde  im Messenger Videochat!



_______________________________________________

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

iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
 		 	   		  
Alles in einem Postfach  ?   Ich will Hotmail!


_______________________________________________

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
 		 	   		  
_________________________________________________________________
http://redirect.gimas.net/?n=M1003xIMVideochat2
Treffe Freunde im Messenger Videochat!
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From anthony.minessale at gmail.com  Fri Mar 26 09:31:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 11:31:43 -0500
Subject: [Freeswitch-dev] hooked functions are not called while using
	one single session
In-Reply-To: <SNT133-w55A5A011A53DC01B04C610A0230@phx.gbl>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>
	<191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>
	<SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>
	<191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>
	<SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>
	<191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>
	<SNT133-w324431DD601582973573FEA0270@phx.gbl>
	<191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>
	<SNT133-w55A5A011A53DC01B04C610A0230@phx.gbl>
Message-ID: <191c3a031003260931k2f10e495q1ad1e0b7e76eaf19@mail.gmail.com>

you can only write back the exact number of bytes you negotiated per
interval.


On Fri, Mar 26, 2010 at 7:27 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:

>  Now I call switch_core_session_read_frame and
> switch_core_session_write_frame in the same foreground loop. Then I write
> audio into a file and as a result I see that there are many audio gaps.
>
> There seems to be a cross effect by using switch_core_session_write_frame
> and switch_core_session_read_frame together.
> If I only use switch_core_session_read_frame in the loop, there are no gaps
> and everything sounds perfect!
>
> What could be the problem?
>
>
>
> while (1)
> {
>     // READ
>     tstatus = switch_core_session_read_frame(session, &read_frame,
> SWITCH_IO_FLAG_NONE, 0);
>     if (tstatus==SWITCH_STATUS_SUCCESS)
>     {        // write audio in a testfile
>             fwrite((short *)(read_frame)->data, 2,
> ((read_frame)->datalen)/2 ,tmp);
>             switch_size_t ts_rf=read_frame->timestamp;
>     }
>
>     // WRITE
>     vector<unsigned char> vucIn;
>     if (getAudio(&vucAudio)) // get next audio and send it back
>     {
>         memcpy( write_frame.data, &vucIn[0], vucIn.size());
>         write_frame.datalen=vucIn.size();
>         tstatus = switch_core_session_write_frame(session, &write_frame,
> SWITCH_IO_FLAG_NONE, 0);
>     }
>     //Sleep(1);//?
> }
>
> ------------------------------
> Date: Mon, 22 Mar 2010 18:34:46 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you don't use the hook functions you use a frame pointer
>
> switch_frame_t *read_frame;
>
> switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE,
> 0);
>
> when i said to look at the stream file example i meant in the function
> itself to see how
> to read and write audio from a channel in a foreground loop.
>
>
>
> On Mon, Mar 22, 2010 at 6:23 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  I have an selfmade software which is controlled by a telephone call
> (ISDN/CAPI). The user calls with his phone, gets
> connected and can speak commands which are detected by an internal speech
> recognizer. Furthermore the user can accept
> or hangup incoming calls just by his voice command. It is also possible to
> dial out.
>
> Now I am adding VOIP functionality to my system and decided to use
> FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP
> interface for my software. The module must be able to:
>
> 1. recognize an incoming call which is the users control session (single
> session)
>    I do this in statehandler function channel_on_init.
> 2. pick up another incoming call and intercept with my users control
> session.
> 3. Get the audio data from voip session (user control session) and send it
> to my application (by own tcp-socket)
>     My idea is to use channel_write_frame to receive audio.
> 3. Send audio data from my application to voip user control session so the
> user can hear audio.
>     My idea is to use channel_read_frame.
>
> explanation for using hooks:
> ----------------------------
> The endpoint interface callback functions where never called in my module,
> even I tried to adapt the way it is done
> in mod_portaudio. So my workaround was to set hook functions
> channel_read_frame/channel_write_frame to handle audio.
>
> My problems:
> ------------
> The hook functions are working fine if I have 2 sessions (put together by
> switch_ivr_intercept_session), but if I only have the single user control
> session no hooked function is called. In this case the only way to achieve
> that hooks are called, is to play audio with switch_ivr_play_file. But then,
> audio samples got lost sometimes.
>
> Maybe there is a better way to solve my requirements?
>
>
>
>
> ------------------------------
> Date: Mon, 22 Mar 2010 11:55:42 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> Why do you even need the hooks?
> What exactly are you trying to do?
> Maybe you should explain it.
>
>
> On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  I am now using the application interface and play a file after answering
> the call. The call is not connected to a another session, it's just a single
> session.
>
> SWITCH_STANDARD_APP(mod_my_function)
> {
>      /* play audio file */
>     switch_ivr_play_file(session, NULL,
> "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav",
> NULL);
> }
>
> Now the hooks for channel_write_frame and channel_read_frame are called! I
> write the audio frames into a file for testing if everything is correct. The
> data written in channel_write_frame seems to be perfect, but audio written
> in channel_read_frame only seems to be good at first sight. If I record 20
> seconds of audio and listen to the audiofile I hear that 2 or 3 times short
> parts are missing.
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>   // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into
> a file
>   fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
>
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t **frame, switch_io_flag_t flags, int stream_id)
> {
>     // Spoken audio samples from my microphone are written, but sometimes
> samples are missing
>     fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
> }
>
> Is it possible that I don't get all audio samples by hooked functions? why
> audio recorded in channel_write_frame is brilliant and in channel_read_frame
> not?
>
>
>
>
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 13:59:28 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you probably want to use the application interface and a while loop while
> reading a frame.
>
> see most of the functions in switch_ivr_play_say
>
>
> On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  The module I am writing should be an endpoint VOIP-interface for another
> program. The module should answer a call and transport incoming audio into a
> 3rd party application. Furthermore the 3rd party application sends audio
> data back to the caller. This is why I hooked channel_read_frame and
> channel_write_frame.
>
> How can I trigger the channel to do something that requires reading or
> writing audio? Does it make a difference to use the endpoint interface
> instead of the hooks?
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 08:34:18 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
>
> What exactly is your goal?
>
> those hooks will probably only be called if you send the channel to do
> something that
> requires reading and writing audio.
>
>
> On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  Hello,
> I am writing my own freeswitch module and have implemented 2 hooks for
> reading and writing audio (I did not implement the endpoint interface).
> My problem is that the channel_write_frame and channel_read_frame
> callback-functions are only called if I make an intercept with another
> session (switch_ivr_intercept_session), but my usecase has only one single
> Session for getting and putting audio data in my module.
>
> How can I achieve this?
> scaram
>
> /* on channel init I answer the session and set hooks */
> static switch_status_t channel_on_init(switch_core_session_t *session)
> {
>     switch_channel_t *channel = switch_core_session_get_channel(session);
>     switch_channel_answer(channel);
>
>     switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
>     switch_set_flag(tech_pvt, TFLAG_IO);
>
>     switch_core_event_hook_add_write_frame(session, &channel_write_frame);
>     switch_core_event_hook_add_read_frame(session, &channel_read_frame);
> }
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
>
>
>
> ------------------------------
> Ein Postfach f?r Alles ?   bei Hotmail 5 E-Mail-Adressen online verwalten!<http://redirect.gimas.net/?n=M1003HM5Adressen>
>
> _______________________________________________
> 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
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
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Twitter: http://twitter.com/FreeSWITCH_wire

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From scaram at hotmail.de  Fri Mar 26 10:23:06 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Fri, 26 Mar 2010 18:23:06 +0100
Subject: [Freeswitch-dev] hooked functions are not called while
	using	one single session
In-Reply-To: <191c3a031003260931k2f10e495q1ad1e0b7e76eaf19@mail.gmail.com>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>,
	<191c3a031003160634vbc3ce61ka3f919e4bfaddd64@mail.gmail.com>,
	<SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>,
	<191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>,
	<SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>,
	<191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>,
	<SNT133-w324431DD601582973573FEA0270@phx.gbl>,
	<191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>,
	<SNT133-w55A5A011A53DC01B04C610A0230@phx.gbl>,
	<191c3a031003260931k2f10e495q1ad1e0b7e76eaf19@mail.gmail.com>
Message-ID: <SNT133-w377811CC94108A22C47B3AA0230@phx.gbl>


ok,  which is the best mechanism to controll the amount of data. Can you recommend a timer? I use PCMA codec with 20 msec intervall.


Date: Fri, 26 Mar 2010 11:31:43 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you can only write back the exact number of bytes you negotiated per interval.


On Fri, Mar 26, 2010 at 7:27 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:






Now I call switch_core_session_read_frame and switch_core_session_write_frame in the same foreground loop. Then I write audio into a file and as a result I see that there are many audio gaps. 

There seems to be a cross effect by using switch_core_session_write_frame and switch_core_session_read_frame together.

If I only use switch_core_session_read_frame in the loop, there are no gaps and everything sounds perfect!

What could be the problem? 



while (1)
{
    // READ
    tstatus = switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);

    if (tstatus==SWITCH_STATUS_SUCCESS)
    {        // write audio in a testfile
            fwrite((short *)(read_frame)->data, 2, ((read_frame)->datalen)/2 ,tmp);
            switch_size_t ts_rf=read_frame->timestamp;

    }

    // WRITE
    vector<unsigned char> vucIn;
    if (getAudio(&vucAudio)) // get next audio and send it back
    {
        memcpy( write_frame.data, &vucIn[0], vucIn.size());
        write_frame.datalen=vucIn.size();

        tstatus = switch_core_session_write_frame(session, &write_frame, SWITCH_IO_FLAG_NONE, 0);
    }
    //Sleep(1);//?
}

Date: Mon, 22 Mar 2010 18:34:46 -0500

From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session


you don't use the hook functions you use a frame pointer 

switch_frame_t *read_frame;

switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);

when i said to look at the stream file example i meant in the function itself to see how 


to read and write audio from a channel in a foreground loop.



On Mon, Mar 22, 2010 at 6:23 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:







I have an selfmade software which is controlled by a telephone call (ISDN/CAPI). The user calls with his phone, gets 
connected and can speak commands which are detected by an internal speech recognizer. Furthermore the user can accept 


or hangup incoming calls just by his voice command. It is also possible to dial out. 

Now I am adding VOIP functionality to my system and decided to use FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP interface for my software. The module must be able to:



1. recognize an incoming call which is the users control session (single session)
   I do this in statehandler function channel_on_init.
2. pick up another incoming call and intercept with my users control session. 


3. Get the audio data from voip session (user control session) and send it to my application (by own tcp-socket)
    My idea is to use channel_write_frame to receive audio.
3. Send audio data from my application to voip user control session so the user can hear audio.


    My idea is to use channel_read_frame.

explanation for using hooks:
----------------------------
The endpoint interface callback functions where never called in my module, even I tried to adapt the way it is done 


in mod_portaudio. So my workaround was to set hook functions channel_read_frame/channel_write_frame to handle audio. 

My problems:
------------
The hook functions are working fine if I have 2 sessions (put together by switch_ivr_intercept_session), but if I only have the single user control session no hooked function is called. In this case the only way to achieve that hooks are called, is to play audio with switch_ivr_play_file. But then, audio samples got lost sometimes.



Maybe there is a better way to solve my requirements? 




Date: Mon, 22 Mar 2010 11:55:42 -0500
From: anthony.minessale at gmail.com


To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

Why do you even need the hooks?


What exactly are you trying to do?
Maybe you should explain it.


On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:








I am now using the application interface and play a file after answering the call. The call is not connected to a another session, it's just a single session.

SWITCH_STANDARD_APP(mod_my_function)
{
     /* play audio file */



    switch_ivr_play_file(session, NULL, "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav", NULL);
}

Now the hooks for channel_write_frame and channel_read_frame are called! I write the audio frames into a file for testing if everything is correct. The data written in channel_write_frame seems to be perfect, but audio written in channel_read_frame only seems to be good at first sight. If I record 20 seconds of audio and listen to the audiofile I hear that 2 or 3 times short parts are missing. 




static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
  // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into a file



  fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t **frame, switch_io_flag_t flags, int stream_id)

{
    // Spoken audio samples from my microphone are written, but sometimes samples are missing
    fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}

Is it possible that I don't get all audio samples by hooked functions? why audio recorded in channel_write_frame is brilliant and in channel_read_frame not?









Date: Tue, 16 Mar 2010 13:59:28 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org



Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you probably want to use the application interface and a while loop while reading a frame.

see most of the functions in switch_ivr_play_say





On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:






The module I am writing should be an endpoint VOIP-interface for another program. The module should answer a call and transport incoming audio into a 3rd party application. Furthermore the 3rd party application sends audio data back to the caller. This is why I hooked channel_read_frame and channel_write_frame. 





How can I trigger the channel to do something that requires reading or writing audio? Does it make a difference to use the endpoint interface instead of the hooks? 


Date: Tue, 16 Mar 2010 08:34:18 -0500




From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session





What exactly is your goal?

those hooks will probably only be called if you send the channel to do something that 
requires reading and writing audio.


On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:










Hello,
I am writing my own freeswitch module and have implemented 2 hooks for reading and writing audio (I did not implement the endpoint interface).
My problem is that the channel_write_frame and channel_read_frame callback-functions are only called if I make an intercept with another session (switch_ivr_intercept_session), but my usecase has only one single Session for getting and putting audio data in my module. 






How can I achieve this? 
scaram

/* on channel init I answer the session and set hooks */
static switch_status_t channel_on_init(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);





    switch_channel_answer(channel);

    switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
    switch_set_flag(tech_pvt, TFLAG_IO);

    switch_core_event_hook_add_write_frame(session, &channel_write_frame);





    switch_core_event_hook_add_read_frame(session, &channel_read_frame);
}

static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)





{
    /* is only called when I intercept 2 Sessions */
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
    /* is only called when I intercept 2 Sessions */





}


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




Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com



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



Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com


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


Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com

GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

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

Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

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From anthony.minessale at gmail.com  Fri Mar 26 10:37:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 12:37:04 -0500
Subject: [Freeswitch-dev] hooked functions are not called while using
	one single session
In-Reply-To: <SNT133-w377811CC94108A22C47B3AA0230@phx.gbl>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>
	<SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>
	<191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>
	<SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>
	<191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>
	<SNT133-w324431DD601582973573FEA0270@phx.gbl>
	<191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>
	<SNT133-w55A5A011A53DC01B04C610A0230@phx.gbl>
	<191c3a031003260931k2f10e495q1ad1e0b7e76eaf19@mail.gmail.com>
	<SNT133-w377811CC94108A22C47B3AA0230@phx.gbl>
Message-ID: <191c3a031003261037r27e53b58qfd3a46051f730370@mail.gmail.com>

You can use the data you have,

the read will already be timed right

when you get a frame that has flag SFF_CNG you know that there was not a
real frame to read but you should still write
the size id determined from getting the read_impl and using
read_impl->encoded_bytes_per_frame



On Fri, Mar 26, 2010 at 12:23 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:

>  ok,  which is the best mechanism to controll the amount of data. Can you
> recommend a timer? I use PCMA codec with 20 msec intervall.
>
>
> ------------------------------
> Date: Fri, 26 Mar 2010 11:31:43 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you can only write back the exact number of bytes you negotiated per
> interval.
>
>
> On Fri, Mar 26, 2010 at 7:27 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  Now I call switch_core_session_read_frame and
> switch_core_session_write_frame in the same foreground loop. Then I write
> audio into a file and as a result I see that there are many audio gaps.
>
> There seems to be a cross effect by using switch_core_session_write_frame
> and switch_core_session_read_frame together.
> If I only use switch_core_session_read_frame in the loop, there are no gaps
> and everything sounds perfect!
>
> What could be the problem?
>
>
>
> while (1)
> {
>     // READ
>     tstatus = switch_core_session_read_frame(session, &read_frame,
> SWITCH_IO_FLAG_NONE, 0);
>     if (tstatus==SWITCH_STATUS_SUCCESS)
>     {        // write audio in a testfile
>             fwrite((short *)(read_frame)->data, 2,
> ((read_frame)->datalen)/2 ,tmp);
>             switch_size_t ts_rf=read_frame->timestamp;
>     }
>
>     // WRITE
>     vector<unsigned char> vucIn;
>     if (getAudio(&vucAudio)) // get next audio and send it back
>     {
>         memcpy( write_frame.data, &vucIn[0], vucIn.size());
>         write_frame.datalen=vucIn.size();
>         tstatus = switch_core_session_write_frame(session, &write_frame,
> SWITCH_IO_FLAG_NONE, 0);
>     }
>     //Sleep(1);//?
> }
>
> ------------------------------
> Date: Mon, 22 Mar 2010 18:34:46 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you don't use the hook functions you use a frame pointer
>
> switch_frame_t *read_frame;
>
> switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE,
> 0);
>
> when i said to look at the stream file example i meant in the function
> itself to see how
> to read and write audio from a channel in a foreground loop.
>
>
>
> On Mon, Mar 22, 2010 at 6:23 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  I have an selfmade software which is controlled by a telephone call
> (ISDN/CAPI). The user calls with his phone, gets
> connected and can speak commands which are detected by an internal speech
> recognizer. Furthermore the user can accept
> or hangup incoming calls just by his voice command. It is also possible to
> dial out.
>
> Now I am adding VOIP functionality to my system and decided to use
> FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP
> interface for my software. The module must be able to:
>
> 1. recognize an incoming call which is the users control session (single
> session)
>    I do this in statehandler function channel_on_init.
> 2. pick up another incoming call and intercept with my users control
> session.
> 3. Get the audio data from voip session (user control session) and send it
> to my application (by own tcp-socket)
>     My idea is to use channel_write_frame to receive audio.
> 3. Send audio data from my application to voip user control session so the
> user can hear audio.
>     My idea is to use channel_read_frame.
>
> explanation for using hooks:
> ----------------------------
> The endpoint interface callback functions where never called in my module,
> even I tried to adapt the way it is done
> in mod_portaudio. So my workaround was to set hook functions
> channel_read_frame/channel_write_frame to handle audio.
>
> My problems:
> ------------
> The hook functions are working fine if I have 2 sessions (put together by
> switch_ivr_intercept_session), but if I only have the single user control
> session no hooked function is called. In this case the only way to achieve
> that hooks are called, is to play audio with switch_ivr_play_file. But then,
> audio samples got lost sometimes.
>
> Maybe there is a better way to solve my requirements?
>
>
>
>
> ------------------------------
> Date: Mon, 22 Mar 2010 11:55:42 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> Why do you even need the hooks?
> What exactly are you trying to do?
> Maybe you should explain it.
>
>
> On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  I am now using the application interface and play a file after answering
> the call. The call is not connected to a another session, it's just a single
> session.
>
> SWITCH_STANDARD_APP(mod_my_function)
> {
>      /* play audio file */
>     switch_ivr_play_file(session, NULL,
> "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav",
> NULL);
> }
>
> Now the hooks for channel_write_frame and channel_read_frame are called! I
> write the audio frames into a file for testing if everything is correct. The
> data written in channel_write_frame seems to be perfect, but audio written
> in channel_read_frame only seems to be good at first sight. If I record 20
> seconds of audio and listen to the audiofile I hear that 2 or 3 times short
> parts are missing.
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>   // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into
> a file
>   fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
>
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t **frame, switch_io_flag_t flags, int stream_id)
> {
>     // Spoken audio samples from my microphone are written, but sometimes
> samples are missing
>     fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
> }
>
> Is it possible that I don't get all audio samples by hooked functions? why
> audio recorded in channel_write_frame is brilliant and in channel_read_frame
> not?
>
>
>
>
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 13:59:28 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you probably want to use the application interface and a while loop while
> reading a frame.
>
> see most of the functions in switch_ivr_play_say
>
>
> On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  The module I am writing should be an endpoint VOIP-interface for another
> program. The module should answer a call and transport incoming audio into a
> 3rd party application. Furthermore the 3rd party application sends audio
> data back to the caller. This is why I hooked channel_read_frame and
> channel_write_frame.
>
> How can I trigger the channel to do something that requires reading or
> writing audio? Does it make a difference to use the endpoint interface
> instead of the hooks?
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 08:34:18 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
>
> What exactly is your goal?
>
> those hooks will probably only be called if you send the channel to do
> something that
> requires reading and writing audio.
>
>
> On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  Hello,
> I am writing my own freeswitch module and have implemented 2 hooks for
> reading and writing audio (I did not implement the endpoint interface).
> My problem is that the channel_write_frame and channel_read_frame
> callback-functions are only called if I make an intercept with another
> session (switch_ivr_intercept_session), but my usecase has only one single
> Session for getting and putting audio data in my module.
>
> How can I achieve this?
> scaram
>
> /* on channel init I answer the session and set hooks */
> static switch_status_t channel_on_init(switch_core_session_t *session)
> {
>     switch_channel_t *channel = switch_core_session_get_channel(session);
>     switch_channel_answer(channel);
>
>     switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
>     switch_set_flag(tech_pvt, TFLAG_IO);
>
>     switch_core_event_hook_add_write_frame(session, &channel_write_frame);
>     switch_core_event_hook_add_read_frame(session, &channel_read_frame);
> }
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
>
>
>
> ------------------------------
> Ein Postfach f?r Alles ?   bei Hotmail 5 E-Mail-Adressen online verwalten!<http://redirect.gimas.net/?n=M1003HM5Adressen>
>
> _______________________________________________
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> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
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> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
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> FreeSWITCH-dev at lists.freeswitch.org
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> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Treffe Freunde im Messenger Videochat!<http://redirect.gimas.net/?n=M1003IMVideochat>
>
> _______________________________________________
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> 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
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
> 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<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Treffe Freunde im Messenger Videochat!<http://redirect.gimas.net/?n=M1003IMVideochat>
>
> _______________________________________________
> 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<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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-------------- next part --------------
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From scaram at hotmail.de  Fri Mar 26 10:55:04 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Fri, 26 Mar 2010 18:55:04 +0100
Subject: [Freeswitch-dev] hooked functions are not called while
	using	one single session
In-Reply-To: <191c3a031003261037r27e53b58qfd3a46051f730370@mail.gmail.com>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>,
	<SNT133-w51F2A84E2B2C0FECFDD494A02D0@phx.gbl>,
	<191c3a031003161159x734dc1c1y94a9079218b014e6@mail.gmail.com>,
	<SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>,
	<191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>,
	<SNT133-w324431DD601582973573FEA0270@phx.gbl>,
	<191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>,
	<SNT133-w55A5A011A53DC01B04C610A0230@phx.gbl>,
	<191c3a031003260931k2f10e495q1ad1e0b7e76eaf19@mail.gmail.com>,
	<SNT133-w377811CC94108A22C47B3AA0230@phx.gbl>,
	<191c3a031003261037r27e53b58qfd3a46051f730370@mail.gmail.com>
Message-ID: <SNT133-w36DF971B966B882126F3B8A0230@phx.gbl>


yes, switch_core_session_read_frame will be timed right. This is working for me, but my troubles are in switch_core_session_write_frame.
Is this function timed write too when I put data into this function?

Date: Fri, 26 Mar 2010 12:37:04 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

You can use the data you have,

yes the the read will already be timed right 

when you get a frame that has flag SFF_CNG you know that there was not a real frame to read but you should still write
the size id determined from getting the read_impl and using read_impl->encoded_bytes_per_frame




On Fri, Mar 26, 2010 at 12:23 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:






ok,  which is the best mechanism to controll the amount of data. Can you recommend a timer? I use PCMA codec with 20 msec intervall.


Date: Fri, 26 Mar 2010 11:31:43 -0500

From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session


you can only write back the exact number of bytes you negotiated per interval.


On Fri, Mar 26, 2010 at 7:27 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:







Now I call switch_core_session_read_frame and switch_core_session_write_frame in the same foreground loop. Then I write audio into a file and as a result I see that there are many audio gaps. 

There seems to be a cross effect by using switch_core_session_write_frame and switch_core_session_read_frame together.


If I only use switch_core_session_read_frame in the loop, there are no gaps and everything sounds perfect!

What could be the problem? 



while (1)
{
    // READ
    tstatus = switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);


    if (tstatus==SWITCH_STATUS_SUCCESS)
    {        // write audio in a testfile
            fwrite((short *)(read_frame)->data, 2, ((read_frame)->datalen)/2 ,tmp);
            switch_size_t ts_rf=read_frame->timestamp;


    }

    // WRITE
    vector<unsigned char> vucIn;
    if (getAudio(&vucAudio)) // get next audio and send it back
    {
        memcpy( write_frame.data, &vucIn[0], vucIn.size());
        write_frame.datalen=vucIn.size();


        tstatus = switch_core_session_write_frame(session, &write_frame, SWITCH_IO_FLAG_NONE, 0);
    }
    //Sleep(1);//?
}

Date: Mon, 22 Mar 2010 18:34:46 -0500

From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session



you don't use the hook functions you use a frame pointer 

switch_frame_t *read_frame;

switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);

when i said to look at the stream file example i meant in the function itself to see how 



to read and write audio from a channel in a foreground loop.



On Mon, Mar 22, 2010 at 6:23 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:








I have an selfmade software which is controlled by a telephone call (ISDN/CAPI). The user calls with his phone, gets 
connected and can speak commands which are detected by an internal speech recognizer. Furthermore the user can accept 



or hangup incoming calls just by his voice command. It is also possible to dial out. 

Now I am adding VOIP functionality to my system and decided to use FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP interface for my software. The module must be able to:




1. recognize an incoming call which is the users control session (single session)
   I do this in statehandler function channel_on_init.
2. pick up another incoming call and intercept with my users control session. 



3. Get the audio data from voip session (user control session) and send it to my application (by own tcp-socket)
    My idea is to use channel_write_frame to receive audio.
3. Send audio data from my application to voip user control session so the user can hear audio.



    My idea is to use channel_read_frame.

explanation for using hooks:
----------------------------
The endpoint interface callback functions where never called in my module, even I tried to adapt the way it is done 



in mod_portaudio. So my workaround was to set hook functions channel_read_frame/channel_write_frame to handle audio. 

My problems:
------------
The hook functions are working fine if I have 2 sessions (put together by switch_ivr_intercept_session), but if I only have the single user control session no hooked function is called. In this case the only way to achieve that hooks are called, is to play audio with switch_ivr_play_file. But then, audio samples got lost sometimes.




Maybe there is a better way to solve my requirements? 




Date: Mon, 22 Mar 2010 11:55:42 -0500
From: anthony.minessale at gmail.com



To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

Why do you even need the hooks?



What exactly are you trying to do?
Maybe you should explain it.


On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:









I am now using the application interface and play a file after answering the call. The call is not connected to a another session, it's just a single session.

SWITCH_STANDARD_APP(mod_my_function)
{
     /* play audio file */




    switch_ivr_play_file(session, NULL, "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav", NULL);
}

Now the hooks for channel_write_frame and channel_read_frame are called! I write the audio frames into a file for testing if everything is correct. The data written in channel_write_frame seems to be perfect, but audio written in channel_read_frame only seems to be good at first sight. If I record 20 seconds of audio and listen to the audiofile I hear that 2 or 3 times short parts are missing. 





static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
  // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into a file




  fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t **frame, switch_io_flag_t flags, int stream_id)

{
    // Spoken audio samples from my microphone are written, but sometimes samples are missing
    fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}

Is it possible that I don't get all audio samples by hooked functions? why audio recorded in channel_write_frame is brilliant and in channel_read_frame not?










Date: Tue, 16 Mar 2010 13:59:28 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org




Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you probably want to use the application interface and a while loop while reading a frame.

see most of the functions in switch_ivr_play_say






On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:






The module I am writing should be an endpoint VOIP-interface for another program. The module should answer a call and transport incoming audio into a 3rd party application. Furthermore the 3rd party application sends audio data back to the caller. This is why I hooked channel_read_frame and channel_write_frame. 






How can I trigger the channel to do something that requires reading or writing audio? Does it make a difference to use the endpoint interface instead of the hooks? 


Date: Tue, 16 Mar 2010 08:34:18 -0500





From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session






What exactly is your goal?

those hooks will probably only be called if you send the channel to do something that 
requires reading and writing audio.


On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:











Hello,
I am writing my own freeswitch module and have implemented 2 hooks for reading and writing audio (I did not implement the endpoint interface).
My problem is that the channel_write_frame and channel_read_frame callback-functions are only called if I make an intercept with another session (switch_ivr_intercept_session), but my usecase has only one single Session for getting and putting audio data in my module. 







How can I achieve this? 
scaram

/* on channel init I answer the session and set hooks */
static switch_status_t channel_on_init(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);






    switch_channel_answer(channel);

    switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
    switch_set_flag(tech_pvt, TFLAG_IO);

    switch_core_event_hook_add_write_frame(session, &channel_write_frame);






    switch_core_event_hook_add_read_frame(session, &channel_read_frame);
}

static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)






{
    /* is only called when I intercept 2 Sessions */
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
    /* is only called when I intercept 2 Sessions */






}


 		 	   		  
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ClueCon http://www.cluecon.com/






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





Twitter: http://twitter.com/FreeSWITCH_wire

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Twitter: http://twitter.com/FreeSWITCH_wire

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


Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com

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

Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
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From mike at jerris.com  Fri Mar 26 10:59:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Mar 2010 13:59:17 -0400
Subject: [Freeswitch-dev] Attention all FreeSWITCH and Openzap committors.
	ACTION REQUIRED
Message-ID: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>

We are in process of our migration from svn to git for our main source control.  We will likely still have a read only svn mirror, but all future commits will go to git after the migration.  In order to proceed with this process, I will need your full name, public key, email address, and svn username.  If you have outstanding svn branches that you would like to keep, please contact me with these details as well, I do not currently plan on merging any branches from svn to the new git repo.

For information on how to use git: http://git-scm.com/documentation , http://git-scm.com/course/svn.html
For information on how to generate a public key if you do not already have one:  http://hkn.eecs.berkeley.edu/~dhsu/ssh_public_key_howto.html

For general users, more information will follow on these changes.  For now, nothing has changed.  Around April 1, there will likely be a 1 time need to do a full checkout if you plan on continuing to use the svn mirror, or a fresh git checkout if you plan on switching to git.

If you have any other questions about the process or changes, feel free to ask.

Mike



From anthony.minessale at gmail.com  Fri Mar 26 11:07:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 13:07:14 -0500
Subject: [Freeswitch-dev] hooked functions are not called while using
	one single session
In-Reply-To: <SNT133-w36DF971B966B882126F3B8A0230@phx.gbl>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>
	<SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>
	<191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>
	<SNT133-w324431DD601582973573FEA0270@phx.gbl>
	<191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>
	<SNT133-w55A5A011A53DC01B04C610A0230@phx.gbl>
	<191c3a031003260931k2f10e495q1ad1e0b7e76eaf19@mail.gmail.com>
	<SNT133-w377811CC94108A22C47B3AA0230@phx.gbl>
	<191c3a031003261037r27e53b58qfd3a46051f730370@mail.gmail.com>
	<SNT133-w36DF971B966B882126F3B8A0230@phx.gbl>
Message-ID: <191c3a031003261107y794010b4u3d434ba856e3e61e@mail.gmail.com>

you put 1 frame in write for each read with the data size i told you about
in the last email


On Fri, Mar 26, 2010 at 12:55 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:

>  yes, switch_core_session_read_frame will be timed right. This is working
> for me, but my troubles are in switch_core_session_write_frame.
> Is this function timed write too when I put data into this function?
>
> ------------------------------
> Date: Fri, 26 Mar 2010 12:37:04 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> You can use the data you have,
>
> yes the the read will already be timed right
>
> when you get a frame that has flag SFF_CNG you know that there was not a
> real frame to read but you should still write
> the size id determined from getting the read_impl and using
> read_impl->encoded_bytes_per_frame
>
>
>
> On Fri, Mar 26, 2010 at 12:23 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  ok,  which is the best mechanism to controll the amount of data. Can you
> recommend a timer? I use PCMA codec with 20 msec intervall.
>
>
> ------------------------------
> Date: Fri, 26 Mar 2010 11:31:43 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you can only write back the exact number of bytes you negotiated per
> interval.
>
>
> On Fri, Mar 26, 2010 at 7:27 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  Now I call switch_core_session_read_frame and
> switch_core_session_write_frame in the same foreground loop. Then I write
> audio into a file and as a result I see that there are many audio gaps.
>
> There seems to be a cross effect by using switch_core_session_write_frame
> and switch_core_session_read_frame together.
> If I only use switch_core_session_read_frame in the loop, there are no gaps
> and everything sounds perfect!
>
> What could be the problem?
>
>
>
> while (1)
> {
>     // READ
>     tstatus = switch_core_session_read_frame(session, &read_frame,
> SWITCH_IO_FLAG_NONE, 0);
>     if (tstatus==SWITCH_STATUS_SUCCESS)
>     {        // write audio in a testfile
>             fwrite((short *)(read_frame)->data, 2,
> ((read_frame)->datalen)/2 ,tmp);
>             switch_size_t ts_rf=read_frame->timestamp;
>     }
>
>     // WRITE
>     vector<unsigned char> vucIn;
>     if (getAudio(&vucAudio)) // get next audio and send it back
>     {
>         memcpy( write_frame.data, &vucIn[0], vucIn.size());
>         write_frame.datalen=vucIn.size();
>         tstatus = switch_core_session_write_frame(session, &write_frame,
> SWITCH_IO_FLAG_NONE, 0);
>     }
>     //Sleep(1);//?
> }
>
> ------------------------------
> Date: Mon, 22 Mar 2010 18:34:46 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you don't use the hook functions you use a frame pointer
>
> switch_frame_t *read_frame;
>
> switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE,
> 0);
>
> when i said to look at the stream file example i meant in the function
> itself to see how
> to read and write audio from a channel in a foreground loop.
>
>
>
> On Mon, Mar 22, 2010 at 6:23 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  I have an selfmade software which is controlled by a telephone call
> (ISDN/CAPI). The user calls with his phone, gets
> connected and can speak commands which are detected by an internal speech
> recognizer. Furthermore the user can accept
> or hangup incoming calls just by his voice command. It is also possible to
> dial out.
>
> Now I am adding VOIP functionality to my system and decided to use
> FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP
> interface for my software. The module must be able to:
>
> 1. recognize an incoming call which is the users control session (single
> session)
>    I do this in statehandler function channel_on_init.
> 2. pick up another incoming call and intercept with my users control
> session.
> 3. Get the audio data from voip session (user control session) and send it
> to my application (by own tcp-socket)
>     My idea is to use channel_write_frame to receive audio.
> 3. Send audio data from my application to voip user control session so the
> user can hear audio.
>     My idea is to use channel_read_frame.
>
> explanation for using hooks:
> ----------------------------
> The endpoint interface callback functions where never called in my module,
> even I tried to adapt the way it is done
> in mod_portaudio. So my workaround was to set hook functions
> channel_read_frame/channel_write_frame to handle audio.
>
> My problems:
> ------------
> The hook functions are working fine if I have 2 sessions (put together by
> switch_ivr_intercept_session), but if I only have the single user control
> session no hooked function is called. In this case the only way to achieve
> that hooks are called, is to play audio with switch_ivr_play_file. But then,
> audio samples got lost sometimes.
>
> Maybe there is a better way to solve my requirements?
>
>
>
>
> ------------------------------
> Date: Mon, 22 Mar 2010 11:55:42 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> Why do you even need the hooks?
> What exactly are you trying to do?
> Maybe you should explain it.
>
>
> On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  I am now using the application interface and play a file after answering
> the call. The call is not connected to a another session, it's just a single
> session.
>
> SWITCH_STANDARD_APP(mod_my_function)
> {
>      /* play audio file */
>     switch_ivr_play_file(session, NULL,
> "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav",
> NULL);
> }
>
> Now the hooks for channel_write_frame and channel_read_frame are called! I
> write the audio frames into a file for testing if everything is correct. The
> data written in channel_write_frame seems to be perfect, but audio written
> in channel_read_frame only seems to be good at first sight. If I record 20
> seconds of audio and listen to the audiofile I hear that 2 or 3 times short
> parts are missing.
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>   // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into
> a file
>   fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
>
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t **frame, switch_io_flag_t flags, int stream_id)
> {
>     // Spoken audio samples from my microphone are written, but sometimes
> samples are missing
>     fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
> }
>
> Is it possible that I don't get all audio samples by hooked functions? why
> audio recorded in channel_write_frame is brilliant and in channel_read_frame
> not?
>
>
>
>
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 13:59:28 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you probably want to use the application interface and a while loop while
> reading a frame.
>
> see most of the functions in switch_ivr_play_say
>
>
> On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  The module I am writing should be an endpoint VOIP-interface for another
> program. The module should answer a call and transport incoming audio into a
> 3rd party application. Furthermore the 3rd party application sends audio
> data back to the caller. This is why I hooked channel_read_frame and
> channel_write_frame.
>
> How can I trigger the channel to do something that requires reading or
> writing audio? Does it make a difference to use the endpoint interface
> instead of the hooks?
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 08:34:18 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
>
> What exactly is your goal?
>
> those hooks will probably only be called if you send the channel to do
> something that
> requires reading and writing audio.
>
>
> On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  Hello,
> I am writing my own freeswitch module and have implemented 2 hooks for
> reading and writing audio (I did not implement the endpoint interface).
> My problem is that the channel_write_frame and channel_read_frame
> callback-functions are only called if I make an intercept with another
> session (switch_ivr_intercept_session), but my usecase has only one single
> Session for getting and putting audio data in my module.
>
> How can I achieve this?
> scaram
>
> /* on channel init I answer the session and set hooks */
> static switch_status_t channel_on_init(switch_core_session_t *session)
> {
>     switch_channel_t *channel = switch_core_session_get_channel(session);
>     switch_channel_answer(channel);
>
>     switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
>     switch_set_flag(tech_pvt, TFLAG_IO);
>
>     switch_core_event_hook_add_write_frame(session, &channel_write_frame);
>     switch_core_event_hook_add_read_frame(session, &channel_read_frame);
> }
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
>
>
>
> ------------------------------
> Ein Postfach f?r Alles ?   bei Hotmail 5 E-Mail-Adressen online verwalten!<http://redirect.gimas.net/?n=M1003HM5Adressen>
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> _______________________________________________
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>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
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> FreeSWITCH-dev at lists.freeswitch.org
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>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Treffe Freunde im Messenger Videochat!<http://redirect.gimas.net/?n=M1003IMVideochat>
>
> _______________________________________________
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> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888<http://conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
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>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Treffe Freunde im Messenger Videochat!<http://redirect.gimas.net/?n=M1003IMVideochat>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
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>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Alles in einem Postfach ? Ich will Hotmail!<http://redirect.gimas.net/?n=M1003Hotmail>
>
> _______________________________________________
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> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
> Damit Privates privat bleibt! Internet Explorer 8!<http://redirect.gimas.net/?n=M1003IE8>
>
> _______________________________________________
> 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 <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From msc at freeswitch.org  Fri Mar 26 11:18:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Mar 2010 11:18:11 -0700
Subject: [Freeswitch-dev] FreeSWITCH Pub Conf is bridged to the VUC
	conference
Message-ID: <87f2f3b91003261118h14541896j57b5602744f39088@mail.gmail.com>

Feel free to call in to the FS public conference to listen in as the FS team
and the VUC gang talk tech!

SIP: 888 at conference.freeswitch.org
PSTN: +1-919-386-9900

VUC IRC channel: #vuc

Come join us!
-MC
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From javieraristizabal at gmail.com  Fri Mar 26 14:59:12 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Fri, 26 Mar 2010 16:59:12 -0500
Subject: [Freeswitch-dev] Attention all FreeSWITCH and Openzap
	committors. ACTION REQUIRED
In-Reply-To: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>
References: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>
Message-ID: <b6f11f301003261459l5f5606efjacfc64f9847e9733@mail.gmail.com>

Hi Michael.

My username is: javar
I attached my ssh_key.

Many thanks.

-.-- --- / ... --- -.--

Javier Aristiz?bal
Senior Developer

On Fri, Mar 26, 2010 at 12:59 PM, Michael Jerris <mike at jerris.com> wrote:

> We are in process of our migration from svn to git for our main source
> control.  We will likely still have a read only svn mirror, but all future
> commits will go to git after the migration.  In order to proceed with this
> process, I will need your full name, public key, email address, and svn
> username.  If you have outstanding svn branches that you would like to keep,
> please contact me with these details as well, I do not currently plan on
> merging any branches from svn to the new git repo.
>
> For information on how to use git: http://git-scm.com/documentation ,
> http://git-scm.com/course/svn.html
> For information on how to generate a public key if you do not already have
> one:  http://hkn.eecs.berkeley.edu/~dhsu/ssh_public_key_howto.html
>
> For general users, more information will follow on these changes.  For now,
> nothing has changed.  Around April 1, there will likely be a 1 time need to
> do a full checkout if you plan on continuing to use the svn mirror, or a
> fresh git checkout if you plan on switching to git.
>
> If you have any other questions about the process or changes, feel free to
> ask.
>
> Mike
>
>
> _______________________________________________
> 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 scaram at hotmail.de  Fri Mar 26 17:26:24 2010
From: scaram at hotmail.de (Francisco Scaramanga)
Date: Sat, 27 Mar 2010 01:26:24 +0100
Subject: [Freeswitch-dev] hooked functions are not called while
	using	one single session
In-Reply-To: <191c3a031003261107y794010b4u3d434ba856e3e61e@mail.gmail.com>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>,
	<SNT133-w57C3C9FA1D8A0AED80EFB7A0270@phx.gbl>,
	<191c3a031003220955g1af09b31u523a51167ea7e34b@mail.gmail.com>,
	<SNT133-w324431DD601582973573FEA0270@phx.gbl>,
	<191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>,
	<SNT133-w55A5A011A53DC01B04C610A0230@phx.gbl>,
	<191c3a031003260931k2f10e495q1ad1e0b7e76eaf19@mail.gmail.com>,
	<SNT133-w377811CC94108A22C47B3AA0230@phx.gbl>,
	<191c3a031003261037r27e53b58qfd3a46051f730370@mail.gmail.com>,
	<SNT133-w36DF971B966B882126F3B8A0230@phx.gbl>,
	<191c3a031003261107y794010b4u3d434ba856e3e61e@mail.gmail.com>
Message-ID: <SNT133-w202C90975089CB4AE4C743A0220@phx.gbl>


Function switch_ivr_session_echo is doing the same thing as I do in my test, and it seems that something doesn't work correctly.
If you just edit switch_ivr_session_echo and add 3 lines of code for logging audio data into a file, you can listen to the audio gaps. 

To reproduce the gaps it's the best to read a text for about 60 seconds, then you will hear it.

For the test I edited the dialplan:

 <extension name="testit22">
      <condition field="destination_number" expression="47112">
         <action application="answer"/>
         <action application="echo"/>
      </condition>
  </extension>
    
.. and the code..

SWITCH_DECLARE(void) switch_ivr_session_echo(switch_core_session_t *session, switch_input_args_t *args)
{
        FILE *tmp = NULL;
...
...
        status = switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);
        if (!SWITCH_READ_ACCEPTABLE(status)) {
            break;
        }
        /*******************************************************************************************************************/
        /*      TEST CODE FOR WRITING AUDIO INTO FILE , IF RECORD 1 MINUTE AND LISTEN TO THE FILE YOU WILL GET AUDIO GAPS. */
                tmp = fopen("C:\\files\\recorded_echo", "ab");
                fwrite((short *)(read_frame)->data, 2, ((read_frame)->datalen)/2 ,tmp);
                fclose(tmp);
        /*******************************************************************************************************************/
...
...
}

Furthermore, If you remove the line "switch_core_session_write_frame(session, read_frame, SWITCH_IO_FLAG_NONE, 0);" in function switch_ivr_session_echo, echo will not be sent back, but then spoken audio will be recorded correctly. This is what I meant with possible cross effect.

Date: Fri, 26 Mar 2010 13:07:14 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you put 1 frame in write for each read with the data size i told you about in the last email


On Fri, Mar 26, 2010 at 12:55 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:






yes, switch_core_session_read_frame will be timed right. This is working for me, but my troubles are in switch_core_session_write_frame.
Is this function timed write too when I put data into this function?


Date: Fri, 26 Mar 2010 12:37:04 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org

Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

You can use the data you have,

yes the the read will already be timed right 


when you get a frame that has flag SFF_CNG you know that there was not a real frame to read but you should still write
the size id determined from getting the read_impl and using read_impl->encoded_bytes_per_frame





On Fri, Mar 26, 2010 at 12:23 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:







ok,  which is the best mechanism to controll the amount of data. Can you recommend a timer? I use PCMA codec with 20 msec intervall.


Date: Fri, 26 Mar 2010 11:31:43 -0500

From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session



you can only write back the exact number of bytes you negotiated per interval.


On Fri, Mar 26, 2010 at 7:27 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:








Now I call switch_core_session_read_frame and switch_core_session_write_frame in the same foreground loop. Then I write audio into a file and as a result I see that there are many audio gaps. 

There seems to be a cross effect by using switch_core_session_write_frame and switch_core_session_read_frame together.



If I only use switch_core_session_read_frame in the loop, there are no gaps and everything sounds perfect!

What could be the problem? 



while (1)
{
    // READ
    tstatus = switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);



    if (tstatus==SWITCH_STATUS_SUCCESS)
    {        // write audio in a testfile
            fwrite((short *)(read_frame)->data, 2, ((read_frame)->datalen)/2 ,tmp);
            switch_size_t ts_rf=read_frame->timestamp;



    }

    // WRITE
    vector<unsigned char> vucIn;
    if (getAudio(&vucAudio)) // get next audio and send it back
    {
        memcpy( write_frame.data, &vucIn[0], vucIn.size());
        write_frame.datalen=vucIn.size();



        tstatus = switch_core_session_write_frame(session, &write_frame, SWITCH_IO_FLAG_NONE, 0);
    }
    //Sleep(1);//?
}

Date: Mon, 22 Mar 2010 18:34:46 -0500

From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session




you don't use the hook functions you use a frame pointer 

switch_frame_t *read_frame;

switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE, 0);

when i said to look at the stream file example i meant in the function itself to see how 




to read and write audio from a channel in a foreground loop.



On Mon, Mar 22, 2010 at 6:23 PM, Francisco Scaramanga <scaram at hotmail.de> wrote:









I have an selfmade software which is controlled by a telephone call (ISDN/CAPI). The user calls with his phone, gets 
connected and can speak commands which are detected by an internal speech recognizer. Furthermore the user can accept 




or hangup incoming calls just by his voice command. It is also possible to dial out. 

Now I am adding VOIP functionality to my system and decided to use FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP interface for my software. The module must be able to:





1. recognize an incoming call which is the users control session (single session)
   I do this in statehandler function channel_on_init.
2. pick up another incoming call and intercept with my users control session. 




3. Get the audio data from voip session (user control session) and send it to my application (by own tcp-socket)
    My idea is to use channel_write_frame to receive audio.
3. Send audio data from my application to voip user control session so the user can hear audio.




    My idea is to use channel_read_frame.

explanation for using hooks:
----------------------------
The endpoint interface callback functions where never called in my module, even I tried to adapt the way it is done 




in mod_portaudio. So my workaround was to set hook functions channel_read_frame/channel_write_frame to handle audio. 

My problems:
------------
The hook functions are working fine if I have 2 sessions (put together by switch_ivr_intercept_session), but if I only have the single user control session no hooked function is called. In this case the only way to achieve that hooks are called, is to play audio with switch_ivr_play_file. But then, audio samples got lost sometimes.





Maybe there is a better way to solve my requirements? 




Date: Mon, 22 Mar 2010 11:55:42 -0500
From: anthony.minessale at gmail.com




To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

Why do you even need the hooks?




What exactly are you trying to do?
Maybe you should explain it.


On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:










I am now using the application interface and play a file after answering the call. The call is not connected to a another session, it's just a single session.

SWITCH_STANDARD_APP(mod_my_function)
{
     /* play audio file */





    switch_ivr_play_file(session, NULL, "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav", NULL);
}

Now the hooks for channel_write_frame and channel_read_frame are called! I write the audio frames into a file for testing if everything is correct. The data written in channel_write_frame seems to be perfect, but audio written in channel_read_frame only seems to be good at first sight. If I record 20 seconds of audio and listen to the audiofile I hear that 2 or 3 times short parts are missing. 






static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
  // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into a file





  fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t **frame, switch_io_flag_t flags, int stream_id)

{
    // Spoken audio samples from my microphone are written, but sometimes samples are missing
    fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
}

Is it possible that I don't get all audio samples by hooked functions? why audio recorded in channel_write_frame is brilliant and in channel_read_frame not?











Date: Tue, 16 Mar 2010 13:59:28 -0500
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org





Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session

you probably want to use the application interface and a while loop while reading a frame.

see most of the functions in switch_ivr_play_say







On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:






The module I am writing should be an endpoint VOIP-interface for another program. The module should answer a call and transport incoming audio into a 3rd party application. Furthermore the 3rd party application sends audio data back to the caller. This is why I hooked channel_read_frame and channel_write_frame. 







How can I trigger the channel to do something that requires reading or writing audio? Does it make a difference to use the endpoint interface instead of the hooks? 


Date: Tue, 16 Mar 2010 08:34:18 -0500






From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] hooked functions are not called while using	one single session







What exactly is your goal?

those hooks will probably only be called if you send the channel to do something that 
requires reading and writing audio.


On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de> wrote:












Hello,
I am writing my own freeswitch module and have implemented 2 hooks for reading and writing audio (I did not implement the endpoint interface).
My problem is that the channel_write_frame and channel_read_frame callback-functions are only called if I make an intercept with another session (switch_ivr_intercept_session), but my usecase has only one single Session for getting and putting audio data in my module. 








How can I achieve this? 
scaram

/* on channel init I answer the session and set hooks */
static switch_status_t channel_on_init(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);







    switch_channel_answer(channel);

    switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
    switch_set_flag(tech_pvt, TFLAG_IO);

    switch_core_event_hook_add_write_frame(session, &channel_write_frame);







    switch_core_event_hook_add_read_frame(session, &channel_read_frame);
}

static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)







{
    /* is only called when I intercept 2 Sessions */
}
static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
{
    /* is only called when I intercept 2 Sessions */







}


 		 	   		  
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-- 
Anthony Minessale II

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







Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com






GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org






iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
 		 	   		  
Alles in einem Postfach  ?   Ich will Hotmail!







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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





iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
 		 	   		  
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http://www.freeswitch.org




-- 
Anthony Minessale II

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





Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com




GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org




iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
 		 	   		  
Alles in einem Postfach  ?   Ich will Hotmail!





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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
 		 	   		  
Treffe Freunde  im Messenger Videochat!

_______________________________________________

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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
 		 	   		  
Alles in einem Postfach  ?   Ich will Hotmail!

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http://www.freeswitch.org




-- 
Anthony Minessale II

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


Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com

GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org

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

Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

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From anthony.minessale at gmail.com  Fri Mar 26 17:47:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 19:47:01 -0500
Subject: [Freeswitch-dev] hooked functions are not called while using
	one single session
In-Reply-To: <SNT133-w202C90975089CB4AE4C743A0220@phx.gbl>
References: <SNT133-w58F873FA869C2B18DED818A02D0@phx.gbl>
	<SNT133-w324431DD601582973573FEA0270@phx.gbl>
	<191c3a031003221634j134ae4f8q7d7df1714b2fbff6@mail.gmail.com>
	<SNT133-w55A5A011A53DC01B04C610A0230@phx.gbl>
	<191c3a031003260931k2f10e495q1ad1e0b7e76eaf19@mail.gmail.com>
	<SNT133-w377811CC94108A22C47B3AA0230@phx.gbl>
	<191c3a031003261037r27e53b58qfd3a46051f730370@mail.gmail.com>
	<SNT133-w36DF971B966B882126F3B8A0230@phx.gbl>
	<191c3a031003261107y794010b4u3d434ba856e3e61e@mail.gmail.com>
	<SNT133-w202C90975089CB4AE4C743A0220@phx.gbl>
Message-ID: <191c3a031003261747q76feb4d5p7e44d80988010c64@mail.gmail.com>

The code you are looking at for playback and record has also changed the
format of the channel to signed linear before writing, you are tyring to
mimic that same thing without setting the read codec.

you should be treating the input as unsigned char and writing the full
frame->datalen into your file.

you also may want to not open and close the file every 20ms since that can't
be good on the OS.

I hate to break it to you, but if you look around at all the code in FS,
it's safe to assume that the code to read and write media is well tested,
the more you start understanding that you need to learn more instead of
thinking something is wrong with the code, the sooner you will find what you
are looking for.

You might want to try irc in #freeswitch-dev or #freeswitch on
irc.freenode.net for a little more help since most people don't check this
list for someone asking to learn how to program audio code with FS.






On Fri, Mar 26, 2010 at 7:26 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:

>  Function switch_ivr_session_echo is doing the same thing as I do in my
> test, and it seems that something doesn't work correctly.
> If you just edit switch_ivr_session_echo and add 3 lines of code for
> logging audio data into a file, you can listen to the audio gaps.
>
> To reproduce the gaps it's the best to read a text for about 60 seconds,
> then you will hear it.
>
> For the test I edited the dialplan:
>
>  <extension name="testit22">
>       <condition field="destination_number" expression="47112">
>          <action application="answer"/>
>          <action application="echo"/>
>       </condition>
>   </extension>
>
> .. and the code..
>
> SWITCH_DECLARE(void) switch_ivr_session_echo(switch_core_session_t
> *session, switch_input_args_t *args)
> {
>         FILE *tmp = NULL;
> ...
> ...
>
>         status = switch_core_session_read_frame(session, &read_frame,
> SWITCH_IO_FLAG_NONE, 0);
>         if (!SWITCH_READ_ACCEPTABLE(status)) {
>             break;
>         }
>
> /*******************************************************************************************************************/
>         /*      TEST CODE FOR WRITING AUDIO INTO FILE , IF RECORD 1 MINUTE
> AND LISTEN TO THE FILE YOU WILL GET AUDIO GAPS. */
>                 tmp = fopen("C:\\files\\recorded_echo", "ab");
>
>                 fwrite((short *)(read_frame)->data, 2,
> ((read_frame)->datalen)/2 ,tmp);
>                 fclose(tmp);
>
> /*******************************************************************************************************************/
> ...
> ...
> }
>
> Furthermore, If you remove the line
> "switch_core_session_write_frame(session, read_frame, SWITCH_IO_FLAG_NONE,
> 0);" in function switch_ivr_session_echo, echo will not be sent back, but
> then spoken audio will be recorded correctly. This is what I meant with
> possible cross effect.
>
> ------------------------------
> Date: Fri, 26 Mar 2010 13:07:14 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you put 1 frame in write for each read with the data size i told you about
> in the last email
>
>
> On Fri, Mar 26, 2010 at 12:55 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  yes, switch_core_session_read_frame will be timed right. This is working
> for me, but my troubles are in switch_core_session_write_frame.
> Is this function timed write too when I put data into this function?
>
> ------------------------------
> Date: Fri, 26 Mar 2010 12:37:04 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> You can use the data you have,
>
> yes the the read will already be timed right
>
> when you get a frame that has flag SFF_CNG you know that there was not a
> real frame to read but you should still write
> the size id determined from getting the read_impl and using
> read_impl->encoded_bytes_per_frame
>
>
>
> On Fri, Mar 26, 2010 at 12:23 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  ok,  which is the best mechanism to controll the amount of data. Can you
> recommend a timer? I use PCMA codec with 20 msec intervall.
>
>
> ------------------------------
> Date: Fri, 26 Mar 2010 11:31:43 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you can only write back the exact number of bytes you negotiated per
> interval.
>
>
> On Fri, Mar 26, 2010 at 7:27 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  Now I call switch_core_session_read_frame and
> switch_core_session_write_frame in the same foreground loop. Then I write
> audio into a file and as a result I see that there are many audio gaps.
>
> There seems to be a cross effect by using switch_core_session_write_frame
> and switch_core_session_read_frame together.
> If I only use switch_core_session_read_frame in the loop, there are no gaps
> and everything sounds perfect!
>
> What could be the problem?
>
>
>
> while (1)
> {
>     // READ
>     tstatus = switch_core_session_read_frame(session, &read_frame,
> SWITCH_IO_FLAG_NONE, 0);
>     if (tstatus==SWITCH_STATUS_SUCCESS)
>     {        // write audio in a testfile
>             fwrite((short *)(read_frame)->data, 2,
> ((read_frame)->datalen)/2 ,tmp);
>             switch_size_t ts_rf=read_frame->timestamp;
>     }
>
>     // WRITE
>     vector<unsigned char> vucIn;
>     if (getAudio(&vucAudio)) // get next audio and send it back
>     {
>         memcpy( write_frame.data, &vucIn[0], vucIn.size());
>         write_frame.datalen=vucIn.size();
>         tstatus = switch_core_session_write_frame(session, &write_frame,
> SWITCH_IO_FLAG_NONE, 0);
>     }
>     //Sleep(1);//?
> }
>
> ------------------------------
> Date: Mon, 22 Mar 2010 18:34:46 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you don't use the hook functions you use a frame pointer
>
> switch_frame_t *read_frame;
>
> switch_core_session_read_frame(session, &read_frame, SWITCH_IO_FLAG_NONE,
> 0);
>
> when i said to look at the stream file example i meant in the function
> itself to see how
> to read and write audio from a channel in a foreground loop.
>
>
>
> On Mon, Mar 22, 2010 at 6:23 PM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  I have an selfmade software which is controlled by a telephone call
> (ISDN/CAPI). The user calls with his phone, gets
> connected and can speak commands which are detected by an internal speech
> recognizer. Furthermore the user can accept
> or hangup incoming calls just by his voice command. It is also possible to
> dial out.
>
> Now I am adding VOIP functionality to my system and decided to use
> FreeSwitch. Therefore I wrote a FreeSwitch module which should be the VOIP
> interface for my software. The module must be able to:
>
> 1. recognize an incoming call which is the users control session (single
> session)
>    I do this in statehandler function channel_on_init.
> 2. pick up another incoming call and intercept with my users control
> session.
> 3. Get the audio data from voip session (user control session) and send it
> to my application (by own tcp-socket)
>     My idea is to use channel_write_frame to receive audio.
> 3. Send audio data from my application to voip user control session so the
> user can hear audio.
>     My idea is to use channel_read_frame.
>
> explanation for using hooks:
> ----------------------------
> The endpoint interface callback functions where never called in my module,
> even I tried to adapt the way it is done
> in mod_portaudio. So my workaround was to set hook functions
> channel_read_frame/channel_write_frame to handle audio.
>
> My problems:
> ------------
> The hook functions are working fine if I have 2 sessions (put together by
> switch_ivr_intercept_session), but if I only have the single user control
> session no hooked function is called. In this case the only way to achieve
> that hooks are called, is to play audio with switch_ivr_play_file. But then,
> audio samples got lost sometimes.
>
> Maybe there is a better way to solve my requirements?
>
>
>
>
> ------------------------------
> Date: Mon, 22 Mar 2010 11:55:42 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> Why do you even need the hooks?
> What exactly are you trying to do?
> Maybe you should explain it.
>
>
> On Mon, Mar 22, 2010 at 5:33 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  I am now using the application interface and play a file after answering
> the call. The call is not connected to a another session, it's just a single
> session.
>
> SWITCH_STANDARD_APP(mod_my_function)
> {
>      /* play audio file */
>     switch_ivr_play_file(session, NULL,
> "C:\\freeswitch1.0.4\\debug\\sounds\\music\\8000\\danza-espanola-op-37-h-142-xii-arabesca.wav",
> NULL);
> }
>
> Now the hooks for channel_write_frame and channel_read_frame are called! I
> write the audio frames into a file for testing if everything is correct. The
> data written in channel_write_frame seems to be perfect, but audio written
> in channel_read_frame only seems to be good at first sight. If I record 20
> seconds of audio and listen to the audiofile I hear that 2 or 3 times short
> parts are missing.
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>   // Audio from danza-espanola-op-37-h-142-xii-arabesca.wav is written into
> a file
>   fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
>
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t **frame, switch_io_flag_t flags, int stream_id)
> {
>     // Spoken audio samples from my microphone are written, but sometimes
> samples are missing
>     fwrite((short *)frame->data, 2, frame->datalen/2 ,tmp);
> }
>
> Is it possible that I don't get all audio samples by hooked functions? why
> audio recorded in channel_write_frame is brilliant and in channel_read_frame
> not?
>
>
>
>
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 13:59:28 -0500
>
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
> you probably want to use the application interface and a while loop while
> reading a frame.
>
> see most of the functions in switch_ivr_play_say
>
>
> On Tue, Mar 16, 2010 at 8:51 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  The module I am writing should be an endpoint VOIP-interface for another
> program. The module should answer a call and transport incoming audio into a
> 3rd party application. Furthermore the 3rd party application sends audio
> data back to the caller. This is why I hooked channel_read_frame and
> channel_write_frame.
>
> How can I trigger the channel to do something that requires reading or
> writing audio? Does it make a difference to use the endpoint interface
> instead of the hooks?
>
>
> ------------------------------
> Date: Tue, 16 Mar 2010 08:34:18 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] hooked functions are not called while using
> one single session
>
>
> What exactly is your goal?
>
> those hooks will probably only be called if you send the channel to do
> something that
> requires reading and writing audio.
>
>
> On Tue, Mar 16, 2010 at 6:47 AM, Francisco Scaramanga <scaram at hotmail.de>wrote:
>
>  Hello,
> I am writing my own freeswitch module and have implemented 2 hooks for
> reading and writing audio (I did not implement the endpoint interface).
> My problem is that the channel_write_frame and channel_read_frame
> callback-functions are only called if I make an intercept with another
> session (switch_ivr_intercept_session), but my usecase has only one single
> Session for getting and putting audio data in my module.
>
> How can I achieve this?
> scaram
>
> /* on channel init I answer the session and set hooks */
> static switch_status_t channel_on_init(switch_core_session_t *session)
> {
>     switch_channel_t *channel = switch_core_session_get_channel(session);
>     switch_channel_answer(channel);
>
>     switch_channel_set_state(channel, CS_EXCHANGE_MEDIA);
>     switch_set_flag(tech_pvt, TFLAG_IO);
>
>     switch_core_event_hook_add_write_frame(session, &channel_write_frame);
>     switch_core_event_hook_add_read_frame(session, &channel_read_frame);
> }
>
> static switch_status_t channel_write_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
> static switch_status_t channel_read_frame(switch_core_session_t *session,
> switch_frame_t *frame, switch_io_flag_t flags, int stream_id)
> {
>     /* is only called when I intercept 2 Sessions */
> }
>
>
>
> ------------------------------
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> _______________________________________________
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>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
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>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
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> _______________________________________________
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>
> FreeSWITCH http://www.freeswitch.org/
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>
> FreeSWITCH http://www.freeswitch.org/
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>
> FreeSWITCH http://www.freeswitch.org/
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>
> FreeSWITCH http://www.freeswitch.org/
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From math.parent at gmail.com  Sat Mar 27 02:28:27 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Sat, 27 Mar 2010 10:28:27 +0100
Subject: [Freeswitch-dev] Attention all FreeSWITCH and Openzap
	committors. ACTION REQUIRED
In-Reply-To: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>
References: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>
Message-ID: <960738411003270228o546e3a3bk8360ee68d62d4c4b@mail.gmail.com>

Hello,

On Fri, Mar 26, 2010 at 6:59 PM, Michael Jerris <mike at jerris.com> wrote:
> We are in process of our migration from svn to git for our main source control. ?We will likely still have a read only svn mirror, but all future commits will go to git after the migration. ?In order to proceed with this process, I will need your full name, public key, email address, and svn username. ?If you have outstanding svn branches that you would like to keep, please contact me with these details as well, I do not currently plan on merging any branches from svn to the new git repo.

My login is sathieu
Attached is my key.

(...)

Mathieu Parent
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From andrewkt at aktzero.com  Sat Mar 27 22:32:41 2010
From: andrewkt at aktzero.com (Andrew Thompson)
Date: Sun, 28 Mar 2010 01:32:41 -0400
Subject: [Freeswitch-dev] Moving FreeSWITCH forward (release planning)
Message-ID: <4BAEE9F9.2090307@aktzero.com>

Many of you will skim this and think "I want to see what bkw/anthm/etc 
have to say before I act." Don't. The reason 1.0.5 has not been released 
so far is enough people have not stepped up, took the initiative, and 
done something/anything productive. (If you're actively contributing to 
code/wiki/jiras, I'm probably not talking about you.)

I had a brief, possibly anger inducing chat with anthm on IRC just about 
a week ago. I'm not angry, and I hope he's not angry. What I want is to 
help move FreeSWITCH forward. I realize that by pointing out things that 
need to be done, I volunteer myself to do them. I accept that 
responsibility, and we can deputize anyone else with a pulse who wants 
to own a piece of the process.

I think everyone agrees that 1.0.4 is pretty stale, "archaic" I think 
I've heard. We're now 29 weeks since 1.0.4, the longest we've been 
without a new release since 1.0.0 was released.

I've spent the week studying what release process information I could 
find for other open source projects. I've distilled that as best as I 
can into a temporary wiki page. It may be wrong. It may not be 
appropriate for FreeSWITCH.

But, it's there. If it's horribly wrong, it can be fixed. It can be 
expanded. It can be formalized, agreed upon, and adopted. The packagers 
and developers that come along in the future can look at it and say 'OK, 
this is how they do releases.' They can know when to expect new 
releases, and who to contact for distribution specific issues.

I submit for your review:

http://wiki.freeswitch.org/wiki/User:Tobarja/Release_Process_Improvement

-- 
Andrew Thompson



From Suneel.Papineni at mettoni.com  Wed Mar 31 09:26:45 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Wed, 31 Mar 2010 17:26:45 +0100
Subject: [Freeswitch-dev] Call Hold - moh direction
Message-ID: <3181A30B8C35AB4AA8577B78DDF4613806ADD317@nickel.mettonigroup.com>

Hi,

 

I am testing Call Hold command and got a doubt. Please clarify me if
this is correct.

 

I made a call from extension 1 to FSComm and call is answered. Now I
want to put extension 1 on hold, so call hold command is issued to
FSComm as 

"api uuid_hold <Unique-ID of call leg of extension1>"

This has put call on hold and I expected music on hold to be played to
extension 1, but moh is played to FSComm and not to the extension.
Whereas if I change the command to 

"api uuid_hold <Unique-ID of call leg of FSComm>", then moh is played to
the extension.

 

Clarify me if my expectation is wrong and it is working correctly. I am
testing with FS version "freeswitch-1.0.5-20100321-0400" on Windows
system.

 

Thanks & Regards

Suneel

 


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From janvb at live.com  Wed Mar 31 15:55:16 2010
From: janvb at live.com (Jan Berger)
Date: Thu, 1 Apr 2010 00:55:16 +0200
Subject: [Freeswitch-dev] Start-up Error
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613806ADD317@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF4613806ADD317@nickel.mettonigroup.com>
Message-ID: <SNT135-w461728C108D25DB75F2075D21E0@phx.gbl>


hi,

 

I have installed FreeSWITCH as an service on my laptop, and it normally works fine.

 

But, it fails to start on start-up. All SIP UA's fail because FreeSWITCH start before the laptop register on the wirelss LAN. I see that FreeSWITCH detect that IP is changing, but it still don't bring the SIP server live - I have to manually stop/start it for it to work.

 

Is there any way around this? 

 

Jan
 		 	   		  
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